369 lines
13 KiB
C
369 lines
13 KiB
C
/*
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* This file is part of the Micro Python project, http://micropython.org/
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*
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* The MIT License (MIT)
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*
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* Copyright (c) 2018 Scott Shawcroft for Adafruit Industries
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* 2018 DeanM for Adafruit Industries
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* 2019 Michael Schroeder
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*
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* Permission is hereby granted, free of charge, to any person obtaining a copy
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* of this software and associated documentation files (the "Software"), to deal
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* in the Software without restriction, including without limitation the rights
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* to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
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* copies of the Software, and to permit persons to whom the Software is
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* furnished to do so, subject to the following conditions:
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*
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* The above copyright notice and this permission notice shall be included in
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* all copies or substantial portions of the Software.
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*
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* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
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* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
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* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE
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* AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
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* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
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* OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
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* THE SOFTWARE.
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*/
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#include "shared-bindings/audiomixer/Mixer.h"
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#include "shared-bindings/audiomixer/MixerVoice.h"
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#include <stdint.h>
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#include "py/runtime.h"
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#include "shared-module/audiocore/__init__.h"
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#if defined(__arm__) && __arm__
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#include "cmsis_compiler.h"
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#endif
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void common_hal_audiomixer_mixer_construct(audiomixer_mixer_obj_t *self,
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uint8_t voice_count,
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uint32_t buffer_size,
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uint8_t bits_per_sample,
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bool samples_signed,
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uint8_t channel_count,
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uint32_t sample_rate) {
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self->len = buffer_size / 2 / sizeof(uint32_t) * sizeof(uint32_t);
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self->first_buffer = m_malloc(self->len);
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if (self->first_buffer == NULL) {
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common_hal_audiomixer_mixer_deinit(self);
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m_malloc_fail(self->len);
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}
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self->second_buffer = m_malloc(self->len);
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if (self->second_buffer == NULL) {
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common_hal_audiomixer_mixer_deinit(self);
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m_malloc_fail(self->len);
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}
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self->bits_per_sample = bits_per_sample;
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self->samples_signed = samples_signed;
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self->channel_count = channel_count;
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self->sample_rate = sample_rate;
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self->voice_count = voice_count;
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}
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void common_hal_audiomixer_mixer_deinit(audiomixer_mixer_obj_t *self) {
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self->first_buffer = NULL;
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self->second_buffer = NULL;
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}
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bool common_hal_audiomixer_mixer_deinited(audiomixer_mixer_obj_t *self) {
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return self->first_buffer == NULL;
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}
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uint32_t common_hal_audiomixer_mixer_get_sample_rate(audiomixer_mixer_obj_t *self) {
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return self->sample_rate;
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}
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uint8_t common_hal_audiomixer_mixer_get_channel_count(audiomixer_mixer_obj_t *self) {
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return self->channel_count;
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}
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uint8_t common_hal_audiomixer_mixer_get_bits_per_sample(audiomixer_mixer_obj_t *self) {
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return self->bits_per_sample;
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}
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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++) {
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if (common_hal_audiomixer_mixervoice_get_playing(MP_OBJ_TO_PTR(self->voice[v]))) {
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return true;
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}
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}
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return false;
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}
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void audiomixer_mixer_reset_buffer(audiomixer_mixer_obj_t *self,
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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++) {
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common_hal_audiomixer_mixervoice_stop(self->voice[i]);
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}
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}
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__attribute__((always_inline))
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static inline uint32_t add16signed(uint32_t a, uint32_t b) {
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#if (defined(__ARM_ARCH_7EM__) && (__ARM_ARCH_7EM__ == 1))
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return __QADD16(a, b);
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#else
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uint32_t result = 0;
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for (int8_t i = 0; i < 2; i++) {
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int16_t ai = a >> (sizeof(int16_t) * 8 * i);
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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) {
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intermediate = SHRT_MAX;
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} else if (intermediate < SHRT_MIN) {
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intermediate = SHRT_MIN;
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}
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result |= (((uint32_t)intermediate) & 0xffff) << (sizeof(int16_t) * 8 * i);
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}
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return result;
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#endif
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}
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__attribute__((always_inline))
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static inline uint32_t mult16signed(uint32_t val, int32_t mul) {
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#if (defined(__ARM_ARCH_7EM__) && (__ARM_ARCH_7EM__ == 1))
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mul <<= 16;
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int32_t hi, lo;
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enum { bits = 16 }; // saturate to 16 bits
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enum { shift = 15 }; // shift is done automatically
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asm volatile ("smulwb %0, %1, %2" : "=r" (lo) : "r" (mul), "r" (val));
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asm volatile ("smulwt %0, %1, %2" : "=r" (hi) : "r" (mul), "r" (val));
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asm volatile ("ssat %0, %1, %2, asr %3" : "=r" (lo) : "I" (bits), "r" (lo), "I" (shift));
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asm volatile ("ssat %0, %1, %2, asr %3" : "=r" (hi) : "I" (bits), "r" (hi), "I" (shift));
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asm volatile ("pkhbt %0, %1, %2, lsl #16" : "=r" (val) : "r" (lo), "r" (hi)); // pack
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return val;
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#else
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uint32_t result = 0;
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float mod_mul = (float)mul / (float)((1 << 15) - 1);
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for (int8_t i = 0; i < 2; i++) {
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int16_t ai = (val >> (sizeof(uint16_t) * 8 * i));
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int32_t intermediate = (int32_t)(ai * mod_mul);
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if (intermediate > SHRT_MAX) {
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intermediate = SHRT_MAX;
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} else if (intermediate < SHRT_MIN) {
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intermediate = SHRT_MIN;
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}
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intermediate &= 0x0000FFFF;
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result |= (((uint32_t)intermediate)) << (sizeof(int16_t) * 8 * i);
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}
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return result;
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#endif
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}
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static inline uint32_t tounsigned8(uint32_t val) {
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#if (defined(__ARM_ARCH_7EM__) && (__ARM_ARCH_7EM__ == 1))
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return __UADD8(val, 0x80808080);
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#else
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return val ^ 0x80808080;
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#endif
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}
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static inline uint32_t tounsigned16(uint32_t val) {
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#if (defined(__ARM_ARCH_7EM__) && (__ARM_ARCH_7EM__ == 1))
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return __UADD16(val, 0x80008000);
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#else
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return val ^ 0x80008000;
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#endif
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}
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static inline uint32_t tosigned16(uint32_t val) {
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#if (defined(__ARM_ARCH_7EM__) && (__ARM_ARCH_7EM__ == 1))
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return __UADD16(val, 0x80008000);
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#else
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return val ^ 0x80008000;
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#endif
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}
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static inline uint32_t unpack8(uint16_t val) {
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return ((val & 0xff00) << 16) | ((val & 0x00ff) << 8);
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}
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static inline uint32_t pack8(uint32_t val) {
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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,
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audiomixer_mixervoice_obj_t *voice, bool voices_active,
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uint32_t *word_buffer, uint32_t length) {
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while (length != 0) {
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if (voice->buffer_length == 0) {
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if (!voice->more_data) {
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if (voice->loop) {
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audiosample_reset_buffer(voice->sample, false, 0);
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} else {
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voice->sample = NULL;
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break;
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}
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}
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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.
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voice->buffer_length /= sizeof(uint32_t);
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voice->more_data = result == GET_BUFFER_MORE_DATA;
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}
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}
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uint32_t n = MIN(voice->buffer_length, length);
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uint32_t *src = voice->remaining_buffer;
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uint16_t level = voice->level;
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// First active voice gets copied over verbatim.
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if (!voices_active) {
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if (MP_LIKELY(self->bits_per_sample == 16)) {
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if (MP_LIKELY(self->samples_signed)) {
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for (uint32_t i = 0; i < n; i++) {
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uint32_t v = src[i];
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word_buffer[i] = mult16signed(v, level);
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}
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} else {
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for (uint32_t i = 0; i < n; i++) {
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uint32_t v = src[i];
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v = tosigned16(v);
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word_buffer[i] = mult16signed(v, level);
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}
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}
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} else {
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uint16_t *hword_buffer = (uint16_t *)word_buffer;
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uint16_t *hsrc = (uint16_t *)src;
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for (uint32_t i = 0; i < n * 2; i++) {
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uint32_t word = unpack8(hsrc[i]);
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if (MP_LIKELY(!self->samples_signed)) {
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word = tosigned16(word);
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}
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word = mult16signed(word, level);
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hword_buffer[i] = pack8(word);
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}
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}
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} else {
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if (MP_LIKELY(self->bits_per_sample == 16)) {
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if (MP_LIKELY(self->samples_signed)) {
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for (uint32_t i = 0; i < n; i++) {
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uint32_t word = src[i];
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word_buffer[i] = add16signed(mult16signed(word, level), word_buffer[i]);
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}
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} else {
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for (uint32_t i = 0; i < n; i++) {
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uint32_t word = src[i];
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word = tosigned16(word);
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word_buffer[i] = add16signed(mult16signed(word, level), word_buffer[i]);
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}
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}
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} else {
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uint16_t *hword_buffer = (uint16_t *)word_buffer;
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uint16_t *hsrc = (uint16_t *)src;
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for (uint32_t i = 0; i < n * 2; i++) {
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uint32_t word = unpack8(hsrc[i]);
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if (MP_LIKELY(!self->samples_signed)) {
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word = tosigned16(word);
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}
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word = mult16signed(word, level);
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word = add16signed(word, unpack8(hword_buffer[i]));
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hword_buffer[i] = pack8(word);
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}
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}
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}
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length -= n;
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word_buffer += n;
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voice->remaining_buffer += n;
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voice->buffer_length -= n;
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}
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if (length && !voices_active) {
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for (uint32_t i = 0; i < length; i++) {
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word_buffer[i] = 0;
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}
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}
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}
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audioio_get_buffer_result_t audiomixer_mixer_get_buffer(audiomixer_mixer_obj_t *self,
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bool single_channel_output,
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uint8_t channel,
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uint8_t **buffer,
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uint32_t *buffer_length) {
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if (!single_channel_output) {
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channel = 0;
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}
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uint32_t channel_read_count = self->left_read_count;
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if (channel == 1) {
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channel_read_count = self->right_read_count;
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}
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*buffer_length = self->len;
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bool need_more_data = self->read_count == channel_read_count;
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if (need_more_data) {
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uint32_t *word_buffer;
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if (self->use_first_buffer) {
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*buffer = (uint8_t *)self->first_buffer;
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word_buffer = self->first_buffer;
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} else {
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*buffer = (uint8_t *)self->second_buffer;
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word_buffer = self->second_buffer;
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}
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self->use_first_buffer = !self->use_first_buffer;
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bool voices_active = false;
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uint32_t length = self->len / sizeof(uint32_t);
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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]);
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if (voice->sample) {
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mix_down_one_voice(self, voice, voices_active, word_buffer, length);
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voices_active = true;
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}
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}
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if (!voices_active) {
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for (uint32_t i = 0; i < length; i++) {
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word_buffer[i] = 0;
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}
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}
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if (!self->samples_signed) {
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if (self->bits_per_sample == 16) {
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for (uint32_t i = 0; i < length; i++) {
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word_buffer[i] = tounsigned16(word_buffer[i]);
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}
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} else {
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for (uint32_t i = 0; i < length; i++) {
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word_buffer[i] = tounsigned8(word_buffer[i]);
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}
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}
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}
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self->read_count += 1;
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} else if (!self->use_first_buffer) {
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*buffer = (uint8_t *)self->first_buffer;
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} else {
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*buffer = (uint8_t *)self->second_buffer;
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}
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if (channel == 0) {
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self->left_read_count += 1;
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} else if (channel == 1) {
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self->right_read_count += 1;
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*buffer = *buffer + self->bits_per_sample / 8;
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}
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return GET_BUFFER_MORE_DATA;
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}
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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,
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uint32_t *max_buffer_length, uint8_t *spacing) {
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*single_buffer = false;
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*samples_signed = self->samples_signed;
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*max_buffer_length = self->len;
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if (single_channel_output) {
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*spacing = self->channel_count;
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} else {
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*spacing = 1;
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}
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}
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