Introduce audioio.Mixer which can mix multiple audio samples

to produce a single sample.

Only works with 16 bit samples on the M4.

Fixes #987
This commit is contained in:
Scott Shawcroft 2018-10-02 18:06:52 -07:00
parent 7b95818c4f
commit 76008ce304
No known key found for this signature in database
GPG Key ID: FD0EDC4B6C53CA59
11 changed files with 880 additions and 78 deletions

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@ -407,6 +407,8 @@ ifneq ($(CHIP_VARIANT),SAMR21G18A)
audioio/__init__.c \
audioio/AudioOut.c
SRC_SHARED_MODULE += \
audioio/__init__.c \
audioio/Mixer.c \
audioio/RawSample.c \
audioio/WaveFile.c
endif

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@ -40,82 +40,6 @@ static audio_dma_t* audio_dma_state[AUDIO_DMA_CHANNEL_COUNT];
// This cannot be in audio_dma_state because it's volatile.
static volatile bool audio_dma_pending[AUDIO_DMA_CHANNEL_COUNT];
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;
}
static 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);
}
}
static audioio_get_buffer_result_t 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 GET_BUFFER_DONE;
}
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++) {

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@ -0,0 +1,249 @@
/*
* 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/Mixer.h"
#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/RawSample.h"
#include "shared-bindings/util.h"
#include "supervisor/shared/translate.h"
//| .. currentmodule:: audioio
//|
//| :class:`Mixer` -- Mixes one or more audio samples together
//| ===========================================================
//|
//| Mixer mixes multiple samples into one sample.
//|
//| .. class:: Mixer(channel_count=2, buffer_size=1024)
//|
//| Create a Mixer object that can mix multiple channels with the same sample rate.
//|
//| :param int channel_count: The maximum number of samples to mix at once
//| :param int buffer_size: The total size in bytes of the buffers to mix into
//|
//| 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)
//|
//| music = audioio.WaveFile(open("cplay-5.1-16bit-16khz.wav", "rb"))
//| drum = audioio.WaveFile(open("drum.wav", "rb"))
//| mixer = audioio.Mixer(voice_count=2, sample_rate=16000, channel_count=1, bits_per_sample=16, samples_signed=True)
//| a = audioio.AudioOut(board.A0)
//|
//| print("playing")
//| a.play(mixer)
//| mixer.play(music, voice=0)
//| while mixer.playing:
//| mixer.play(drum, voice=1)
//| time.sleep(1)
//| print("stopped")
//|
STATIC mp_obj_t audioio_mixer_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, 0, 2, true);
mp_map_t kw_args;
mp_map_init_fixed_table(&kw_args, n_kw, pos_args + n_args);
enum { ARG_voice_count, ARG_buffer_size, ARG_channel_count, ARG_bits_per_sample, ARG_samples_signed, ARG_sample_rate };
static const mp_arg_t allowed_args[] = {
{ MP_QSTR_voice_count, MP_ARG_INT | MP_ARG_KW_ONLY, {.u_int = 2} },
{ MP_QSTR_buffer_size, MP_ARG_INT | MP_ARG_KW_ONLY, {.u_int = 1024} },
{ MP_QSTR_channel_count, MP_ARG_INT | MP_ARG_KW_ONLY, {.u_int = 2} },
{ MP_QSTR_bits_per_sample, MP_ARG_INT | MP_ARG_KW_ONLY, {.u_int = 16} },
{ MP_QSTR_samples_signed, MP_ARG_BOOL | MP_ARG_KW_ONLY, {.u_bool = true} },
{ 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);
mp_int_t voice_count = args[ARG_voice_count].u_int;
if (voice_count < 1 || voice_count > 255) {
mp_raise_ValueError(translate("Invalid voice count"));
}
mp_int_t channel_count = args[ARG_channel_count].u_int;
if (channel_count < 1 || channel_count > 2) {
mp_raise_ValueError(translate("Invalid channel count"));
}
mp_int_t sample_rate = args[ARG_sample_rate].u_int;
if (sample_rate < 1) {
mp_raise_ValueError(translate("Sample rate must be positive"));
}
mp_int_t bits_per_sample = args[ARG_bits_per_sample].u_int;
if (bits_per_sample != 8 && bits_per_sample != 16) {
mp_raise_ValueError(translate("bits_per_sample must be 8 or 16"));
}
audioio_mixer_obj_t *self = m_new_obj_var(audioio_mixer_obj_t, audioio_mixer_voice_t, voice_count);
self->base.type = &audioio_mixer_type;
common_hal_audioio_mixer_construct(self, voice_count, args[ARG_buffer_size].u_int, bits_per_sample, args[ARG_samples_signed].u_bool, channel_count, sample_rate);
return MP_OBJ_FROM_PTR(self);
}
//| .. method:: deinit()
//|
//| Deinitialises the Mixer and releases any hardware resources for reuse.
//|
STATIC mp_obj_t audioio_mixer_deinit(mp_obj_t self_in) {
audioio_mixer_obj_t *self = MP_OBJ_TO_PTR(self_in);
common_hal_audioio_mixer_deinit(self);
return mp_const_none;
}
STATIC MP_DEFINE_CONST_FUN_OBJ_1(audioio_mixer_deinit_obj, audioio_mixer_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_mixer_obj___exit__(size_t n_args, const mp_obj_t *args) {
(void)n_args;
common_hal_audioio_mixer_deinit(args[0]);
return mp_const_none;
}
STATIC MP_DEFINE_CONST_FUN_OBJ_VAR_BETWEEN(audioio_mixer___exit___obj, 4, 4, audioio_mixer_obj___exit__);
//| .. method:: play(sample, *, voice=0, 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`, `audioio.Mixer` or `audioio.RawSample`.
//|
//| If other samples are already playing, the encodings must match.
//|
STATIC mp_obj_t audioio_mixer_obj_play(size_t n_args, const mp_obj_t *pos_args, mp_map_t *kw_args) {
enum { ARG_sample, ARG_voice, ARG_loop };
static const mp_arg_t allowed_args[] = {
{ MP_QSTR_sample, MP_ARG_OBJ | MP_ARG_REQUIRED },
{ MP_QSTR_voice, MP_ARG_INT | MP_ARG_KW_ONLY, {.u_int = 0} },
{ MP_QSTR_loop, MP_ARG_BOOL | MP_ARG_KW_ONLY, {.u_bool = false} },
};
audioio_mixer_obj_t *self = MP_OBJ_TO_PTR(pos_args[0]);
raise_error_if_deinited(common_hal_audioio_mixer_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_audioio_mixer_play(self, sample, args[ARG_voice].u_int, args[ARG_loop].u_bool);
return mp_const_none;
}
MP_DEFINE_CONST_FUN_OBJ_KW(audioio_mixer_play_obj, 1, audioio_mixer_obj_play);
//| .. method:: stop(voice=0)
//|
//| Stops playback and resets to the start of the sample on the given channel.
//|
STATIC mp_obj_t audioio_mixer_obj_stop(size_t n_args, const mp_obj_t *pos_args, mp_map_t *kw_args) {
enum { ARG_voice };
static const mp_arg_t allowed_args[] = {
{ MP_QSTR_voice, MP_ARG_INT | MP_ARG_KW_ONLY, {.u_int = 0} },
};
audioio_mixer_obj_t *self = MP_OBJ_TO_PTR(pos_args[0]);
raise_error_if_deinited(common_hal_audioio_mixer_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_mixer_stop(self, args[ARG_voice].u_int);
return mp_const_none;
}
MP_DEFINE_CONST_FUN_OBJ_KW(audioio_mixer_stop_obj, 1, audioio_mixer_obj_stop);
//| .. attribute:: playing
//|
//| True when an audio sample is being output even if `paused`. (read-only)
//|
STATIC mp_obj_t audioio_mixer_obj_get_playing(mp_obj_t self_in) {
audioio_mixer_obj_t *self = MP_OBJ_TO_PTR(self_in);
raise_error_if_deinited(common_hal_audioio_mixer_deinited(self));
return mp_obj_new_bool(common_hal_audioio_mixer_get_playing(self));
}
MP_DEFINE_CONST_FUN_OBJ_1(audioio_mixer_get_playing_obj, audioio_mixer_obj_get_playing);
const mp_obj_property_t audioio_mixer_playing_obj = {
.base.type = &mp_type_property,
.proxy = {(mp_obj_t)&audioio_mixer_get_playing_obj,
(mp_obj_t)&mp_const_none_obj,
(mp_obj_t)&mp_const_none_obj},
};
//| .. attribute:: sample_rate
//|
//| 32 bit value that dictates how quickly samples are played in Hertz (cycles per second).
//|
STATIC mp_obj_t audioio_mixer_obj_get_sample_rate(mp_obj_t self_in) {
audioio_mixer_obj_t *self = MP_OBJ_TO_PTR(self_in);
raise_error_if_deinited(common_hal_audioio_mixer_deinited(self));
return MP_OBJ_NEW_SMALL_INT(common_hal_audioio_mixer_get_sample_rate(self));
}
MP_DEFINE_CONST_FUN_OBJ_1(audioio_mixer_get_sample_rate_obj, audioio_mixer_obj_get_sample_rate);
const mp_obj_property_t audioio_mixer_sample_rate_obj = {
.base.type = &mp_type_property,
.proxy = {(mp_obj_t)&audioio_mixer_get_sample_rate_obj,
(mp_obj_t)&mp_const_none_obj,
(mp_obj_t)&mp_const_none_obj},
};
STATIC const mp_rom_map_elem_t audioio_mixer_locals_dict_table[] = {
// Methods
{ MP_ROM_QSTR(MP_QSTR_deinit), MP_ROM_PTR(&audioio_mixer_deinit_obj) },
{ MP_ROM_QSTR(MP_QSTR___enter__), MP_ROM_PTR(&default___enter___obj) },
{ MP_ROM_QSTR(MP_QSTR___exit__), MP_ROM_PTR(&audioio_mixer___exit___obj) },
{ MP_ROM_QSTR(MP_QSTR_play), MP_ROM_PTR(&audioio_mixer_play_obj) },
{ MP_ROM_QSTR(MP_QSTR_stop), MP_ROM_PTR(&audioio_mixer_stop_obj) },
// Properties
{ MP_ROM_QSTR(MP_QSTR_playing), MP_ROM_PTR(&audioio_mixer_playing_obj) },
{ MP_ROM_QSTR(MP_QSTR_sample_rate), MP_ROM_PTR(&audioio_mixer_sample_rate_obj) },
};
STATIC MP_DEFINE_CONST_DICT(audioio_mixer_locals_dict, audioio_mixer_locals_dict_table);
const mp_obj_type_t audioio_mixer_type = {
{ &mp_type_type },
.name = MP_QSTR_Mixer,
.make_new = audioio_mixer_make_new,
.locals_dict = (mp_obj_dict_t*)&audioio_mixer_locals_dict,
};

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@ -0,0 +1,52 @@
/*
* 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.
*/
#ifndef MICROPY_INCLUDED_SHARED_BINDINGS_AUDIOIO_MIXER_H
#define MICROPY_INCLUDED_SHARED_BINDINGS_AUDIOIO_MIXER_H
#include "common-hal/microcontroller/Pin.h"
#include "shared-module/audioio/Mixer.h"
#include "shared-bindings/audioio/RawSample.h"
extern const mp_obj_type_t audioio_mixer_type;
void common_hal_audioio_mixer_construct(audioio_mixer_obj_t* self,
uint8_t voice_count,
uint32_t buffer_size,
uint8_t bits_per_sample,
bool samples_signed,
uint8_t channel_count,
uint32_t sample_rate);
void common_hal_audioio_mixer_deinit(audioio_mixer_obj_t* self);
bool common_hal_audioio_mixer_deinited(audioio_mixer_obj_t* self);
void common_hal_audioio_mixer_play(audioio_mixer_obj_t* self, mp_obj_t sample, uint8_t voice, bool loop);
void common_hal_audioio_mixer_stop(audioio_mixer_obj_t* self, uint8_t voice);
bool common_hal_audioio_mixer_get_playing(audioio_mixer_obj_t* self);
uint32_t common_hal_audioio_mixer_get_sample_rate(audioio_mixer_obj_t* self);
#endif // MICROPY_INCLUDED_SHARED_BINDINGS_AUDIOIO_MIXER_H

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@ -32,6 +32,8 @@
#include "shared-bindings/microcontroller/Pin.h"
#include "shared-bindings/audioio/__init__.h"
#include "shared-bindings/audioio/AudioOut.h"
#include "shared-bindings/audioio/Mixer.h"
#include "shared-bindings/audioio/RawSample.h"
#include "shared-bindings/audioio/WaveFile.h"
//| :mod:`audioio` --- Support for audio input and output
@ -49,6 +51,7 @@
//| :maxdepth: 3
//|
//| AudioOut
//| Mixer
//| RawSample
//| WaveFile
//|
@ -61,6 +64,7 @@
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_Mixer), MP_ROM_PTR(&audioio_mixer_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) },
};

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@ -0,0 +1,338 @@
/*
* 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/Mixer.h"
#include <stdint.h>
#include "py/runtime.h"
#include "shared-module/audioio/__init__.h"
#include "shared-module/audioio/RawSample.h"
void common_hal_audioio_mixer_construct(audioio_mixer_obj_t* self,
uint8_t voice_count,
uint32_t buffer_size,
uint8_t bits_per_sample,
bool samples_signed,
uint8_t channel_count,
uint32_t sample_rate) {
self->len = buffer_size / 2 / sizeof(uint32_t) * sizeof(uint32_t);
self->first_buffer = m_malloc(self->len, false);
if (self->first_buffer == NULL) {
common_hal_audioio_mixer_deinit(self);
mp_raise_msg(&mp_type_MemoryError, translate("Couldn't allocate first buffer"));
}
self->second_buffer = m_malloc(self->len, false);
if (self->second_buffer == NULL) {
common_hal_audioio_mixer_deinit(self);
mp_raise_msg(&mp_type_MemoryError, translate("Couldn't allocate second buffer"));
}
self->bits_per_sample = bits_per_sample;
self->samples_signed = samples_signed;
self->channel_count = channel_count;
self->sample_rate = sample_rate;
self->voice_count = voice_count;
for (uint8_t i = 0; i < self->voice_count; i++) {
self->voice[i].sample = NULL;
}
}
void common_hal_audioio_mixer_deinit(audioio_mixer_obj_t* self) {
self->first_buffer = NULL;
self->second_buffer = NULL;
}
bool common_hal_audioio_mixer_deinited(audioio_mixer_obj_t* self) {
return self->first_buffer == NULL;
}
uint32_t common_hal_audioio_mixer_get_sample_rate(audioio_mixer_obj_t* self) {
return self->sample_rate;
}
void common_hal_audioio_mixer_play(audioio_mixer_obj_t* self, mp_obj_t sample, uint8_t v, bool loop) {
if (v >= self->voice_count) {
mp_raise_ValueError(translate("Voice index too high"));
}
if (audiosample_sample_rate(sample) != self->sample_rate) {
mp_raise_ValueError(translate("The sample's sample rate does not match the mixer's"));
}
if (audiosample_channel_count(sample) != self->channel_count) {
mp_raise_ValueError(translate("The sample's channel count does not match the mixer's"));
}
if (audiosample_bits_per_sample(sample) != self->bits_per_sample) {
mp_raise_ValueError(translate("The sample's bits_per_sample does not match the mixer's"));
}
bool single_buffer;
bool samples_signed;
uint32_t max_buffer_length;
uint8_t spacing;
audiosample_get_buffer_structure(sample, false, &single_buffer, &samples_signed,
&max_buffer_length, &spacing);
if (samples_signed != self->samples_signed) {
mp_raise_ValueError(translate("The sample's signedness does not match the mixer's"));
}
audioio_mixer_voice_t* voice = &self->voice[v];
voice->sample = sample;
voice->loop = loop;
audiosample_reset_buffer(sample, false, 0);
audioio_get_buffer_result_t result = audiosample_get_buffer(sample, false, 0, (uint8_t**) &voice->remaining_buffer, &voice->buffer_length);
// Track length in terms of words.
voice->buffer_length /= sizeof(uint32_t);
voice->more_data = result == GET_BUFFER_MORE_DATA;
}
void common_hal_audioio_mixer_stop(audioio_mixer_obj_t* self, uint8_t voice) {
self->voice[voice].sample = NULL;
}
bool common_hal_audioio_mixer_get_playing(audioio_mixer_obj_t* self) {
for (int32_t v = 0; v < self->voice_count; v++) {
if (self->voice[v].sample != NULL) {
return true;
}
}
return false;
}
void audioio_mixer_reset_buffer(audioio_mixer_obj_t* self,
bool single_channel,
uint8_t channel) {
for (int32_t i = 0; i < self->voice_count; i++) {
self->voice[i].sample = NULL;
}
}
uint32_t add8signed(uint32_t a, uint32_t b) {
#if (defined (__ARM_FEATURE_DSP) && (__ARM_FEATURE_DSP == 1))
return __QADD8(a, b);
#else
uint32_t result = 0;
for (int8_t i = 0; i < 4; i++) {
int8_t ai = a >> (sizeof(int8_t) * i);
int8_t bi = b >> (sizeof(int8_t) * i);
int32_t intermediate = (int32_t) ai + bi;
if (intermediate > CHAR_MAX) {
intermediate = CHAR_MAX;
} else if (intermediate < CHAR_MIN) {
intermediate = CHAR_MIN;
}
result |= ((int8_t) intermediate) >> (sizeof(int8_t) * i);
}
return result;
#endif
}
uint32_t add8unsigned(uint32_t a, uint32_t b) {
#if (defined (__ARM_FEATURE_DSP) && (__ARM_FEATURE_DSP == 1))
// Subtract out the DC offset, add and then shift back.
a = __USUB8(a, 0x80808080);
b = __USUB8(b, 0x80808080);
uint32_t sum = __QADD8(a, b);
return __UADD8(sum, 0x80808080);
#else
uint32_t result = 0;
for (int8_t i = 0; i < 4; i++) {
uint8_t ai = a >> (sizeof(uint8_t) * i);
uint8_t bi = b >> (sizeof(uint8_t) * i);
int32_t intermediate = (int32_t) ai + bi;
if (intermediate > UCHAR_MAX) {
intermediate = UCHAR_MAX;
}
result |= ((uint8_t) intermediate) >> (sizeof(uint8_t) * i);
}
return result;
#endif
}
uint32_t add16signed(uint32_t a, uint32_t b) {
#if (defined (__ARM_FEATURE_DSP) && (__ARM_FEATURE_DSP == 1))
return __QADD16(a, b);
#else
uint32_t result = 0;
for (int8_t i = 0; i < 2; i++) {
int16_t ai = a >> (sizeof(int16_t) * i);
int16_t bi = b >> (sizeof(int16_t) * i);
int32_t intermediate = (int32_t) ai + bi;
if (intermediate > SHRT_MAX) {
intermediate = SHRT_MAX;
} else if (intermediate < SHRT_MIN) {
intermediate = SHRT_MIN;
}
result |= ((int16_t) intermediate) >> (sizeof(int16_t) * i);
}
return result;
#endif
}
uint32_t add16unsigned(uint32_t a, uint32_t b) {
#if (defined (__ARM_FEATURE_DSP) && (__ARM_FEATURE_DSP == 1))
// Subtract out the DC offset, add and then shift back.
a = __USUB16(a, 0x80008000);
b = __USUB16(b, 0x80008000);
uint32_t sum = __QADD16(a, b);
return __UADD16(sum, 0x80008000);
#else
uint32_t result = 0;
for (int8_t i = 0; i < 2; i++) {
uint16_t ai = a >> (sizeof(uint16_t) * i);
uint16_t bi = b >> (sizeof(uint16_t) * i);
uint32_t intermediate = (uint32_t) ai + bi;
if (intermediate > USHRT_MAX) {
intermediate = USHRT_MAX;
}
result |= ((uint16_t) intermediate) >> (sizeof(int16_t) * i);
}
return result;
#endif
}
audioio_get_buffer_result_t audioio_mixer_get_buffer(audioio_mixer_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;
}
*buffer_length = self->len;
bool need_more_data = self->read_count == channel_read_count;
if (need_more_data) {
uint32_t* word_buffer;
if (self->use_first_buffer) {
*buffer = (uint8_t*) self->first_buffer;
word_buffer = self->first_buffer;
} else {
*buffer = (uint8_t*) self->second_buffer;
word_buffer = self->second_buffer;
}
self->use_first_buffer = !self->use_first_buffer;
bool voices_active = false;
for (int32_t v = 0; v < self->voice_count; v++) {
audioio_mixer_voice_t* voice = &self->voice[v];
if (voice->sample == NULL) {
continue;
}
uint32_t j = 0;
for (uint32_t i = 0; i < self->len / sizeof(uint32_t); i++) {
if (j >= voice->buffer_length) {
if (!voice->more_data) {
if (voice->loop) {
audiosample_reset_buffer(voice->sample, false, 0);
} else {
voice->sample = NULL;
break;
}
}
// Load another buffer
audioio_get_buffer_result_t result = audiosample_get_buffer(voice->sample, false, 0, (uint8_t**) &voice->remaining_buffer, &voice->buffer_length);
// Track length in terms of words.
voice->buffer_length /= sizeof(uint32_t);
voice->more_data = result == GET_BUFFER_MORE_DATA;
j = 0;
}
// First active voice gets copied over verbatim.
uint32_t sample_value = voice->remaining_buffer[j];
if (!voices_active) {
word_buffer[i] = sample_value;
} else {
if (self->bits_per_sample == 8) {
if (self->samples_signed) {
word_buffer[i] = add8signed(word_buffer[i], sample_value);
} else {
word_buffer[i] = add8unsigned(word_buffer[i], sample_value);
}
} else {
if (self->samples_signed) {
word_buffer[i] = add16signed(word_buffer[i], sample_value);
} else {
word_buffer[i] = add16unsigned(word_buffer[i], sample_value);
}
}
}
j++;
}
voice->buffer_length -= j;
voice->remaining_buffer += j;
voices_active = true;
}
// No voice is active so zero out the signal.
if (!voices_active) {
uint32_t zero = 0;
if (!self->samples_signed) {
if (self->bits_per_sample == 8) {
zero = 0x7f7f7f7f;
} else {
zero = 0x7fff7fff;
}
}
for (uint32_t i = 0; i < self->len / sizeof(uint32_t); i++) {
word_buffer[i] = zero;
}
}
self->read_count += 1;
} else if (!self->use_first_buffer) {
*buffer = (uint8_t*) self->first_buffer;
} else {
*buffer = (uint8_t*) self->second_buffer;
}
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 GET_BUFFER_MORE_DATA;
}
void audioio_mixer_get_buffer_structure(audioio_mixer_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->samples_signed;
*max_buffer_length = self->len;
if (single_channel) {
*spacing = self->channel_count;
} else {
*spacing = 1;
}
}

View File

@ -0,0 +1,75 @@
/*
* 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_MIXER_H
#define MICROPY_INCLUDED_SHARED_MODULE_AUDIOIO_MIXER_H
#include "py/obj.h"
#include "shared-module/audioio/__init__.h"
typedef struct {
mp_obj_t sample;
bool loop;
bool more_data;
uint32_t* remaining_buffer;
uint32_t buffer_length;
} audioio_mixer_voice_t;
typedef struct {
mp_obj_base_t base;
uint32_t* first_buffer;
uint32_t* second_buffer;
uint32_t len; // in words
uint8_t bits_per_sample;
bool use_first_buffer;
bool samples_signed;
uint8_t channel_count;
uint32_t sample_rate;
uint32_t read_count;
uint32_t left_read_count;
uint32_t right_read_count;
uint8_t voice_count;
audioio_mixer_voice_t voice[];
} audioio_mixer_obj_t;
// These are not available from Python because it may be called in an interrupt.
void audioio_mixer_reset_buffer(audioio_mixer_obj_t* self,
bool single_channel,
uint8_t channel);
audioio_get_buffer_result_t audioio_mixer_get_buffer(audioio_mixer_obj_t* self,
bool single_channel,
uint8_t channel,
uint8_t** buffer,
uint32_t* buffer_length); // length in bytes
void audioio_mixer_get_buffer_structure(audioio_mixer_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_MIXER_H

View File

@ -200,13 +200,29 @@ audioio_get_buffer_result_t audioio_wavefile_get_buffer(audioio_wavefile_obj_t*
if (f_read(&self->file->fp, *buffer, num_bytes_to_load, &length_read) != FR_OK) {
return GET_BUFFER_ERROR;
}
self->bytes_remaining -= length_read;
// Pad the last buffer to word align it.
if (self->bytes_remaining == 0 && length_read % sizeof(uint32_t) != 0) {
uint32_t pad = length_read % sizeof(uint32_t);
length_read += pad;
if (self->bits_per_sample == 8) {
for (uint32_t i = 0; i < pad; i++) {
((uint8_t*) (*buffer))[length_read / sizeof(uint8_t) - i - 1] = 0x80;
}
} else if (self->bits_per_sample == 16) {
// We know the buffer is aligned because we allocated it onto the heap ourselves.
#pragma GCC diagnostic push
#pragma GCC diagnostic ignored "-Wcast-align"
((int16_t*) (*buffer))[length_read / sizeof(int16_t) - 1] = 0;
#pragma GCC diagnostic pop
}
}
*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;
}

View File

@ -0,0 +1,125 @@
/*
* 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 "shared-module/audioio/__init__.h"
#include "py/obj.h"
#include "shared-bindings/audioio/Mixer.h"
#include "shared-bindings/audioio/RawSample.h"
#include "shared-bindings/audioio/WaveFile.h"
#include "shared-module/audioio/Mixer.h"
#include "shared-module/audioio/RawSample.h"
#include "shared-module/audioio/WaveFile.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;
} else 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;
} else if (MP_OBJ_IS_TYPE(sample_obj, &audioio_mixer_type)) {
audioio_mixer_obj_t* mixer = MP_OBJ_TO_PTR(sample_obj);
return mixer->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;
} else 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;
} else if (MP_OBJ_IS_TYPE(sample_obj, &audioio_mixer_type)) {
audioio_mixer_obj_t* mixer = MP_OBJ_TO_PTR(sample_obj);
return mixer->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;
} else 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;
} else if (MP_OBJ_IS_TYPE(sample_obj, &audioio_mixer_type)) {
audioio_mixer_obj_t* mixer = MP_OBJ_TO_PTR(sample_obj);
return mixer->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);
} else 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);
} else if (MP_OBJ_IS_TYPE(sample_obj, &audioio_mixer_type)) {
audioio_mixer_obj_t* file = MP_OBJ_TO_PTR(sample_obj);
audioio_mixer_reset_buffer(file, single_channel, audio_channel);
}
}
audioio_get_buffer_result_t 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);
} else 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);
} else if (MP_OBJ_IS_TYPE(sample_obj, &audioio_mixer_type)) {
audioio_mixer_obj_t* file = MP_OBJ_TO_PTR(sample_obj);
return audioio_mixer_get_buffer(file, single_channel, channel, buffer, buffer_length);
}
return GET_BUFFER_DONE;
}
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);
} else if (MP_OBJ_IS_TYPE(sample_obj, &audioio_mixer_type)) {
audioio_mixer_obj_t* file = MP_OBJ_TO_PTR(sample_obj);
audioio_mixer_get_buffer_structure(file, single_channel, single_buffer, samples_signed,
max_buffer_length, spacing);
}
}

View File

@ -27,10 +27,27 @@
#ifndef MICROPY_INCLUDED_SHARED_MODULE_AUDIOIO__INIT__H
#define MICROPY_INCLUDED_SHARED_MODULE_AUDIOIO__INIT__H
#include <stdbool.h>
#include <stdint.h>
#include "py/obj.h"
typedef enum {
GET_BUFFER_DONE, // No more data to read
GET_BUFFER_MORE_DATA, // More data to read.
GET_BUFFER_ERROR, // Error while reading data.
} audioio_get_buffer_result_t;
uint32_t audiosample_sample_rate(mp_obj_t sample_obj);
uint8_t audiosample_bits_per_sample(mp_obj_t sample_obj);
uint8_t audiosample_channel_count(mp_obj_t sample_obj);
void audiosample_reset_buffer(mp_obj_t sample_obj, bool single_channel, uint8_t audio_channel);
audioio_get_buffer_result_t audiosample_get_buffer(mp_obj_t sample_obj,
bool single_channel,
uint8_t channel,
uint8_t** buffer, uint32_t* buffer_length);
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);
#endif // MICROPY_INCLUDED_SHARED_MODULE_AUDIOIO__INIT__H

View File

@ -68,7 +68,7 @@ bool displayio_sprite_get_pixel(displayio_sprite_t *self, int16_t x, int16_t y,
if (y < 0 || y >= self->height || x >= self->width || x < 0) {
return false;
}
uint32_t value;
uint32_t value = 0;
if (MP_OBJ_IS_TYPE(self->bitmap, &displayio_bitmap_type)) {
value = common_hal_displayio_bitmap_get_pixel(self->bitmap, x, y);
} else if (MP_OBJ_IS_TYPE(self->bitmap, &displayio_ondiskbitmap_type)) {