/* * 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 #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_voice(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) * 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) { //intermediate = CHAR_MIN; } 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++) { int8_t ai = (a >> (sizeof(uint8_t) * 8 * i)) - 128; int8_t bi = (b >> (sizeof(uint8_t) * 8 * i)) - 128; int32_t intermediate = (int32_t) ai + bi; if (intermediate > UCHAR_MAX) { intermediate = UCHAR_MAX; } result |= ((uint8_t) intermediate + 128) << (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++) { 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 } 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++) { 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; } result |= ((uint16_t) intermediate + 0x8000) << (sizeof(int16_t) * 8 * 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]; 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]; } 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; } }