circuitpython/shared-module/audiomixer/Mixer.c

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/*
* This file is part of the Micro Python project, http://micropython.org/
*
* The MIT License (MIT)
*
* Copyright (c) 2018 Scott Shawcroft for Adafruit Industries
* 2018 DeanM for Adafruit Industries
* 2019 Michael Schroeder
*
* 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/audiomixer/Mixer.h"
#include "shared-bindings/audiomixer/MixerVoice.h"
#include <stdint.h>
#include "py/runtime.h"
#include "shared-module/audiocore/__init__.h"
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#if defined(__arm__) && __arm__
#include "cmsis_compiler.h"
#endif
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void common_hal_audiomixer_mixer_construct(audiomixer_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);
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self->first_buffer = m_malloc(self->len);
if (self->first_buffer == NULL) {
common_hal_audiomixer_mixer_deinit(self);
m_malloc_fail(self->len);
}
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self->second_buffer = m_malloc(self->len);
if (self->second_buffer == NULL) {
common_hal_audiomixer_mixer_deinit(self);
m_malloc_fail(self->len);
}
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;
}
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void common_hal_audiomixer_mixer_deinit(audiomixer_mixer_obj_t *self) {
self->first_buffer = NULL;
self->second_buffer = NULL;
}
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bool common_hal_audiomixer_mixer_deinited(audiomixer_mixer_obj_t *self) {
return self->first_buffer == NULL;
}
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uint32_t common_hal_audiomixer_mixer_get_sample_rate(audiomixer_mixer_obj_t *self) {
return self->sample_rate;
}
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uint8_t common_hal_audiomixer_mixer_get_channel_count(audiomixer_mixer_obj_t *self) {
return self->channel_count;
}
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uint8_t common_hal_audiomixer_mixer_get_bits_per_sample(audiomixer_mixer_obj_t *self) {
return self->bits_per_sample;
}
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bool common_hal_audiomixer_mixer_get_playing(audiomixer_mixer_obj_t *self) {
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for (uint8_t v = 0; v < self->voice_count; v++) {
if (common_hal_audiomixer_mixervoice_get_playing(MP_OBJ_TO_PTR(self->voice[v]))) {
return true;
}
}
return false;
}
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void audiomixer_mixer_reset_buffer(audiomixer_mixer_obj_t *self,
bool single_channel_output,
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uint8_t channel) {
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for (uint8_t i = 0; i < self->voice_count; i++) {
common_hal_audiomixer_mixervoice_stop(self->voice[i]);
}
}
__attribute__((always_inline))
static inline uint32_t add16signed(uint32_t a, uint32_t b) {
#if (defined(__ARM_ARCH_7EM__) && (__ARM_ARCH_7EM__ == 1))
return __QADD16(a, b);
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#else
uint32_t result = 0;
for (int8_t i = 0; i < 2; i++) {
int16_t ai = a >> (sizeof(int16_t) * 8 * i);
int16_t bi = b >> (sizeof(int16_t) * 8 * i);
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int32_t intermediate = (int32_t)ai + bi;
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if (intermediate > SHRT_MAX) {
intermediate = SHRT_MAX;
} else if (intermediate < SHRT_MIN) {
intermediate = SHRT_MIN;
}
result |= (((uint32_t)intermediate) & 0xffff) << (sizeof(int16_t) * 8 * i);
}
return result;
#endif
}
__attribute__((always_inline))
static inline uint32_t mult16signed(uint32_t val, int32_t mul) {
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#if (defined(__ARM_ARCH_7EM__) && (__ARM_ARCH_7EM__ == 1))
mul <<= 16;
int32_t hi, lo;
enum { bits = 16 }; // saturate to 16 bits
enum { shift = 15 }; // shift is done automatically
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asm volatile ("smulwb %0, %1, %2" : "=r" (lo) : "r" (mul), "r" (val));
asm volatile ("smulwt %0, %1, %2" : "=r" (hi) : "r" (mul), "r" (val));
asm volatile ("ssat %0, %1, %2, asr %3" : "=r" (lo) : "I" (bits), "r" (lo), "I" (shift));
asm volatile ("ssat %0, %1, %2, asr %3" : "=r" (hi) : "I" (bits), "r" (hi), "I" (shift));
asm volatile ("pkhbt %0, %1, %2, lsl #16" : "=r" (val) : "r" (lo), "r" (hi)); // pack
return val;
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#else
uint32_t result = 0;
float mod_mul = (float)mul / (float)((1 << 15) - 1);
for (int8_t i = 0; i < 2; i++) {
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int16_t ai = (val >> (sizeof(uint16_t) * 8 * i));
int32_t intermediate = (int32_t)(ai * mod_mul);
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if (intermediate > SHRT_MAX) {
intermediate = SHRT_MAX;
} else if (intermediate < SHRT_MIN) {
intermediate = SHRT_MIN;
}
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intermediate &= 0x0000FFFF;
result |= (((uint32_t)intermediate)) << (sizeof(int16_t) * 8 * i);
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}
return result;
#endif
}
static inline uint32_t tounsigned8(uint32_t val) {
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#if (defined(__ARM_ARCH_7EM__) && (__ARM_ARCH_7EM__ == 1))
return __UADD8(val, 0x80808080);
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#else
return val ^ 0x80808080;
#endif
}
static inline uint32_t tounsigned16(uint32_t val) {
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#if (defined(__ARM_ARCH_7EM__) && (__ARM_ARCH_7EM__ == 1))
return __UADD16(val, 0x80008000);
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#else
return val ^ 0x80008000;
#endif
}
static inline uint32_t tosigned16(uint32_t val) {
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#if (defined(__ARM_ARCH_7EM__) && (__ARM_ARCH_7EM__ == 1))
return __UADD16(val, 0x80008000);
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#else
return val ^ 0x80008000;
#endif
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}
static inline uint32_t unpack8(uint16_t val) {
return ((val & 0xff00) << 16) | ((val & 0x00ff) << 8);
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}
static inline uint32_t pack8(uint32_t val) {
return ((val & 0xff000000) >> 16) | ((val & 0xff00) >> 8);
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}
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static void mix_down_one_voice(audiomixer_mixer_obj_t *self,
audiomixer_mixervoice_obj_t *voice, bool voices_active,
uint32_t *word_buffer, uint32_t length) {
while (length != 0) {
if (voice->buffer_length == 0) {
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if (!voice->more_data) {
if (voice->loop) {
audiosample_reset_buffer(voice->sample, false, 0);
} else {
voice->sample = NULL;
break;
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}
}
if (voice->sample) {
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// Load another buffer
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audioio_get_buffer_result_t result = audiosample_get_buffer(voice->sample, false, 0, (uint8_t **)&voice->remaining_buffer, &voice->buffer_length);
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// Track length in terms of words.
voice->buffer_length /= sizeof(uint32_t);
voice->more_data = result == GET_BUFFER_MORE_DATA;
}
}
uint32_t n = MIN(voice->buffer_length, length);
uint32_t *src = voice->remaining_buffer;
uint16_t level = voice->level;
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// First active voice gets copied over verbatim.
if (!voices_active) {
if (MP_LIKELY(self->bits_per_sample == 16)) {
if (MP_LIKELY(self->samples_signed)) {
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for (uint32_t i = 0; i < n; i++) {
uint32_t v = src[i];
word_buffer[i] = mult16signed(v, level);
}
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} else {
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for (uint32_t i = 0; i < n; i++) {
uint32_t v = src[i];
v = tosigned16(v);
word_buffer[i] = mult16signed(v, level);
}
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}
} else {
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uint16_t *hword_buffer = (uint16_t *)word_buffer;
uint16_t *hsrc = (uint16_t *)src;
for (uint32_t i = 0; i < n * 2; i++) {
uint32_t word = unpack8(hsrc[i]);
if (MP_LIKELY(!self->samples_signed)) {
word = tosigned16(word);
}
word = mult16signed(word, level);
hword_buffer[i] = pack8(word);
}
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}
} else {
if (MP_LIKELY(self->bits_per_sample == 16)) {
if (MP_LIKELY(self->samples_signed)) {
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for (uint32_t i = 0; i < n; i++) {
uint32_t word = src[i];
word_buffer[i] = add16signed(mult16signed(word, level), word_buffer[i]);
}
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} else {
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for (uint32_t i = 0; i < n; i++) {
uint32_t word = src[i];
word = tosigned16(word);
word_buffer[i] = add16signed(mult16signed(word, level), word_buffer[i]);
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}
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}
} else {
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uint16_t *hword_buffer = (uint16_t *)word_buffer;
uint16_t *hsrc = (uint16_t *)src;
for (uint32_t i = 0; i < n * 2; i++) {
uint32_t word = unpack8(hsrc[i]);
if (MP_LIKELY(!self->samples_signed)) {
word = tosigned16(word);
}
word = mult16signed(word, level);
word = add16signed(word, unpack8(hword_buffer[i]));
hword_buffer[i] = pack8(word);
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}
}
}
length -= n;
word_buffer += n;
voice->remaining_buffer += n;
voice->buffer_length -= n;
}
if (length && !voices_active) {
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for (uint32_t i = 0; i < length; i++) {
word_buffer[i] = 0;
}
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}
}
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audioio_get_buffer_result_t audiomixer_mixer_get_buffer(audiomixer_mixer_obj_t *self,
bool single_channel_output,
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uint8_t channel,
uint8_t **buffer,
uint32_t *buffer_length) {
if (!single_channel_output) {
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) {
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uint32_t *word_buffer;
if (self->use_first_buffer) {
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*buffer = (uint8_t *)self->first_buffer;
word_buffer = self->first_buffer;
} else {
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*buffer = (uint8_t *)self->second_buffer;
word_buffer = self->second_buffer;
}
self->use_first_buffer = !self->use_first_buffer;
bool voices_active = false;
uint32_t length = self->len / sizeof(uint32_t);
for (int32_t v = 0; v < self->voice_count; v++) {
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audiomixer_mixervoice_obj_t *voice = MP_OBJ_TO_PTR(self->voice[v]);
if (voice->sample) {
mix_down_one_voice(self, voice, voices_active, word_buffer, length);
voices_active = true;
}
}
if (!voices_active) {
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for (uint32_t i = 0; i < length; i++) {
word_buffer[i] = 0;
}
}
if (!self->samples_signed) {
if (self->bits_per_sample == 16) {
for (uint32_t i = 0; i < length; i++) {
word_buffer[i] = tounsigned16(word_buffer[i]);
}
} else {
for (uint32_t i = 0; i < length; i++) {
word_buffer[i] = tounsigned8(word_buffer[i]);
}
}
}
self->read_count += 1;
} else if (!self->use_first_buffer) {
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*buffer = (uint8_t *)self->first_buffer;
} else {
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*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 audiomixer_mixer_get_buffer_structure(audiomixer_mixer_obj_t *self, bool single_channel_output,
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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_output) {
*spacing = self->channel_count;
} else {
*spacing = 1;
}
}