circuitpython/shared-module/audiocore/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
*
* 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/audiocore/Mixer.h"
#include "shared-bindings/audiocore/MixerVoice.h"
#include <stdint.h>
#include "py/runtime.h"
#include "shared-module/audiocore/__init__.h"
#include "shared-module/audiocore/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;
}
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;
}
bool common_hal_audioio_mixer_get_playing(audioio_mixer_obj_t* self) {
for (int32_t v = 0; v < self->voice_count; v++) {
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if (common_hal_audioio_mixervoice_get_playing(MP_OBJ_TO_PTR(self->voice[v]))) {
return true;
}
}
return false;
}
void audioio_mixer_reset_buffer(audioio_mixer_obj_t* self,
bool single_channel,
uint8_t channel) {
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#if 0
for (int32_t i = 0; i < self->voice_count; i++) {
self->voice[i].sample = NULL;
}
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#endif
}
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++) {
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int8_t ai = a >> (sizeof(int8_t) * 8 * i);
int8_t bi = b >> (sizeof(int8_t) * 8 * i);
int32_t intermediate = (int32_t) ai + bi;
if (intermediate > CHAR_MAX) {
intermediate = CHAR_MAX;
} else if (intermediate < CHAR_MIN) {
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//intermediate = CHAR_MIN;
}
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result |= (((uint32_t) intermediate) & 0xff) << (sizeof(int8_t) * 8 * 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) * 8 * i));
uint8_t bi = (b >> (sizeof(uint8_t) * 8 * i));
int32_t intermediate = (int32_t) (ai + bi) / 2;
if (intermediate > UCHAR_MAX) {
intermediate = UCHAR_MAX;
}
result |= ((uint32_t) intermediate & 0xff) << (sizeof(uint8_t) * 8 * 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++) {
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int16_t ai = a >> (sizeof(int16_t) * 8 * i);
int16_t bi = b >> (sizeof(int16_t) * 8 * i);
int32_t intermediate = (int32_t) ai + bi;
if (intermediate > SHRT_MAX) {
intermediate = SHRT_MAX;
} else if (intermediate < SHRT_MIN) {
intermediate = SHRT_MIN;
}
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result |= (((uint32_t) intermediate) & 0xffff) << (sizeof(int16_t) * 8 * 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++) {
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int16_t ai = (a >> (sizeof(uint16_t) * 8 * i)) - 0x8000;
int16_t bi = (b >> (sizeof(uint16_t) * 8 * i)) - 0x8000;
int32_t intermediate = (int32_t) ai + bi;
if (intermediate > USHRT_MAX) {
intermediate = USHRT_MAX;
}
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result |= ((uint16_t) intermediate + 0x8000) << (sizeof(int16_t) * 8 * i);
}
return result;
#endif
}
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//TODO:
static inline uint32_t mult8unsigned(uint32_t val, int32_t mul) {
// if mul == 0, no need in wasting cycles
if (mul == 0) {
return 0;
}
#if (defined (__ARM_FEATURE_DSP) && (__ARM_FEATURE_DSP == 1))
int32_t hi, lo;
int32_t bits = 16; // saturate to 16 bits
int32_t shift = 0; // shift is done automatically
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 #8" : "=r" (val) : "r" (lo), "r" (hi)); // pack
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return val;
#else
uint32_t result = 0;
float mod_mul = (float) mul / (float) ((1<<15)-1);
for (int8_t i = 0; i < 4; i++) {
uint8_t ai = val >> (sizeof(uint8_t) * 8 * i);
int32_t intermediate = ai * mod_mul;
if (intermediate > SHRT_MAX) {
intermediate = SHRT_MAX;
}
result |= ((uint32_t) intermediate & 0xff) << (sizeof(uint8_t) * 8 * i);
}
return result;
#endif
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}
//TODO:
static inline uint32_t mult8signed(uint32_t val, int32_t mul) {
// if mul == 0, no need in wasting cycles
if (mul == 0) {
return 0;
}
#if (defined (__ARM_FEATURE_DSP) && (__ARM_FEATURE_DSP == 1))
int32_t hi, lo;
int32_t bits = 16; // saturate to 16 bits
int32_t shift = 0; // shift is done automatically
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 #8" : "=r" (val) : "r" (lo), "r" (hi)); // pack
return val;
#else
uint32_t result = 0;
float mod_mul = (float)mul / (float)((1<<15)-1);
for (int8_t i = 0; i < 4; i++) {
int16_t ai = val >> (sizeof(int8_t) * 8 * i);
int32_t intermediate = ai * mod_mul;
if (intermediate > CHAR_MAX) {
intermediate = CHAR_MAX;
} else if (intermediate < CHAR_MIN) {
intermediate = CHAR_MIN;
}
result |= (((uint32_t) intermediate) & 0xff) << (sizeof(int16_t) * 8 * i);
}
return result;
#endif
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}
//TODO:
static inline uint32_t mult16unsigned(uint32_t val, int32_t mul) {
// if mul == 0, no need in wasting cycles
if (mul == 0) {
return 0;
}
#if (defined (__ARM_FEATURE_DSP) && (__ARM_FEATURE_DSP == 1))
int32_t hi, lo;
asm volatile("umaal %0, %1, %2, %3" : "=r" (lo), "=r" (hi) : "r" (val), "r" (mul)); // unsigned accumulation mult (res=64bit)
asm volatile("pkhbt %0, %1, %2, lsl #16" : "=r" (val) : "r" (lo), "r" (hi)); // pack
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return val;
#else
//mp_printf(&mp_plat_print, "mult16signed called:\n\tval: %u\t mul: %u\n", val, mul);
uint32_t result = 0;
float mod_mul = (float)mul / (float)((1<<15)-1);
for (int8_t i = 0; i < 2; i++) {
int16_t ai = (val >> (sizeof(uint16_t) * 8 * i)) - 0x8000;
int32_t intermediate = ai * mod_mul;
if (intermediate > SHRT_MAX) {
intermediate = SHRT_MAX;
} else if (intermediate < SHRT_MIN) {
intermediate = SHRT_MIN;
}
result |= (((uint32_t) intermediate) + 0x8000) << (sizeof(int16_t) * 8 * i);
}
//mp_printf(&mp_plat_print, "\t mod_mul: %f\t result: %u\n", (double)mod_mul, result);
return val;
#endif
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}
static inline uint32_t mult16signed(uint32_t val, int32_t mul) {
// if mul == 0, no need in wasting cycles
if (mul == 0) {
return 0;
}
#if (defined (__ARM_FEATURE_DSP) && (__ARM_FEATURE_DSP == 1))
int32_t hi, lo;
int32_t bits = 16; // saturate to 16 bits
int32_t shift = 0; // shift is done automatically
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
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return val;
#else
uint32_t result = 0;
float mod_mul = (float)mul / (float)((1<<15)-1);
for (int8_t i = 0; i < 2; i++) {
int16_t ai = val >> (sizeof(int16_t) * 8 * i);
int32_t intermediate = ai * mod_mul;
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
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}
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++) {
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audioio_mixervoice_obj_t* voice = MP_OBJ_TO_PTR(self->voice[v]);
uint32_t j = 0;
bool voice_done = voice->sample == NULL;
for (uint32_t i = 0; i < self->len / sizeof(uint32_t); i++) {
if (!voice_done && j >= voice->buffer_length) {
if (!voice->more_data) {
if (voice->loop) {
audiosample_reset_buffer(voice->sample, false, 0);
} else {
voice->sample = NULL;
voice_done = true;
}
}
if (!voice_done) {
// 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;
if (voice_done) {
// Exit early if another voice already set all samples once.
if (voices_active) {
continue;
}
sample_value = 0;
if (!self->samples_signed) {
if (self->bits_per_sample == 8) {
sample_value = 0x7f7f7f7f;
} else {
sample_value = 0x7fff7fff;
}
}
} else {
sample_value = voice->remaining_buffer[j];
}
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// apply the mixer level
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if (!self->samples_signed) {
if (self->bits_per_sample == 8) {
sample_value = mult8unsigned(sample_value, voice->level);
} else {
sample_value = mult16unsigned(sample_value, voice->level);
}
} else {
if (self->bits_per_sample == 8) {
sample_value = mult8signed(sample_value, voice->level);
} else {
sample_value = mult16signed(sample_value, voice->level);
}
}
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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;
}
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;
}
}