/* * 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 #include "py/runtime.h" #include "shared-module/audiocore/__init__.h" #include "shared-module/audiocore/RawSample.h" 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); self->first_buffer = m_malloc(self->len, false); if (self->first_buffer == NULL) { common_hal_audiomixer_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_audiomixer_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_audiomixer_mixer_deinit(audiomixer_mixer_obj_t *self) { self->first_buffer = NULL; self->second_buffer = NULL; } bool common_hal_audiomixer_mixer_deinited(audiomixer_mixer_obj_t *self) { return self->first_buffer == NULL; } uint32_t common_hal_audiomixer_mixer_get_sample_rate(audiomixer_mixer_obj_t *self) { return self->sample_rate; } uint8_t common_hal_audiomixer_mixer_get_channel_count(audiomixer_mixer_obj_t *self) { return self->channel_count; } uint8_t common_hal_audiomixer_mixer_get_bits_per_sample(audiomixer_mixer_obj_t *self) { return self->bits_per_sample; } bool common_hal_audiomixer_mixer_get_playing(audiomixer_mixer_obj_t *self) { 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; } void audiomixer_mixer_reset_buffer(audiomixer_mixer_obj_t *self, bool single_channel, uint8_t channel) { 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); #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); int32_t intermediate = (int32_t)ai + bi; 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) { #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 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; #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(uint16_t) * 8 * i)); int32_t intermediate = ai * mod_mul; if (intermediate > SHRT_MAX) { intermediate = SHRT_MAX; } else if (intermediate < SHRT_MIN) { intermediate = SHRT_MIN; } intermediate &= 0x0000FFFF; result |= (((uint32_t)intermediate)) << (sizeof(int16_t) * 8 * i); } return result; #endif } static inline uint32_t tounsigned8(uint32_t val) { #if (defined(__ARM_ARCH_7EM__) && (__ARM_ARCH_7EM__ == 1)) return __UADD8(val, 0x80808080); #else return val ^ 0x80808080; #endif } static inline uint32_t tounsigned16(uint32_t val) { #if (defined(__ARM_ARCH_7EM__) && (__ARM_ARCH_7EM__ == 1)) return __UADD16(val, 0x80008000); #else return val ^ 0x80008000; #endif } static inline uint32_t tosigned16(uint32_t val) { #if (defined(__ARM_ARCH_7EM__) && (__ARM_ARCH_7EM__ == 1)) return __UADD16(val, 0x80008000); #else return val ^ 0x80008000; #endif } static inline uint32_t unpack8(uint16_t val) { return ((val & 0xff00) << 16) | ((val & 0x00ff) << 8); } static inline uint32_t pack8(uint32_t val) { return ((val & 0xff000000) >> 16) | ((val & 0xff00) >> 8); } 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) { if (!voice->more_data) { if (voice->loop) { audiosample_reset_buffer(voice->sample, false, 0); } else { voice->sample = NULL; break; } } if (voice->sample) { // 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; } } uint32_t n = MIN(voice->buffer_length, length); uint32_t *src = voice->remaining_buffer; uint16_t level = voice->level; // First active voice gets copied over verbatim. if (!voices_active) { if (MP_LIKELY(self->bits_per_sample == 16)) { if (MP_LIKELY(self->samples_signed)) { for (uint32_t i = 0; i < n; i++) { uint32_t v = src[i]; word_buffer[i] = mult16signed(v, level); } } else { for (uint32_t i = 0; i < n; i++) { uint32_t v = src[i]; v = tosigned16(v); word_buffer[i] = mult16signed(v, level); } } } else { 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); } } } else { if (MP_LIKELY(self->bits_per_sample == 16)) { if (MP_LIKELY(self->samples_signed)) { for (uint32_t i = 0; i < n; i++) { uint32_t word = src[i]; word_buffer[i] = add16signed(mult16signed(word, level), word_buffer[i]); } } else { 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]); } } } else { 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); } } } length -= n; word_buffer += n; voice->remaining_buffer += n; voice->buffer_length -= n; } if (length && !voices_active) { for (uint32_t i = 0; i < length; i++) { word_buffer[i] = 0; } } } audioio_get_buffer_result_t audiomixer_mixer_get_buffer(audiomixer_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; uint32_t length = self->len / sizeof(uint32_t); for (int32_t v = 0; v < self->voice_count; v++) { 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) { 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) { *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 audiomixer_mixer_get_buffer_structure(audiomixer_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; } }