mirror of
https://github.com/eried/portapack-mayhem.git
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5ea1bff1e6
* WAV Viewer & Soundboard enhancements * Reduced width of sample rate field
146 lines
4.7 KiB
C++
146 lines
4.7 KiB
C++
/*
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* Copyright (C) 2015 Jared Boone, ShareBrained Technology, Inc.
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* Copyright (C) 2016 Furrtek
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*
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* This file is part of PortaPack.
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*
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* This program is free software; you can redistribute it and/or modify
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* it under the terms of the GNU General Public License as published by
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* the Free Software Foundation; either version 2, or (at your option)
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* any later version.
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*
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* This program is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
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* GNU General Public License for more details.
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*
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* You should have received a copy of the GNU General Public License
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* along with this program; see the file COPYING. If not, write to
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* the Free Software Foundation, Inc., 51 Franklin Street,
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* Boston, MA 02110-1301, USA.
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*/
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#include "proc_audiotx.hpp"
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#include "portapack_shared_memory.hpp"
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#include "sine_table_int8.hpp"
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#include "event_m4.hpp"
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#include <cstdint>
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void AudioTXProcessor::execute(const buffer_c8_t& buffer) {
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if (!configured) return;
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buffer_s16_t audio_buffer{audio_data, AUDIO_OUTPUT_BUFFER_SIZE, sampling_rate};
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int16_t audio_sample_s16;
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// Zero-order hold (poop)
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for (size_t i = 0; i < buffer.count; i++) {
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resample_acc += resample_inc;
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if (resample_acc >= 0x10000) {
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resample_acc -= 0x10000;
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if (stream) {
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audio_sample = 0;
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stream->read(&audio_sample, bytes_per_sample); // assumes little endian when reading 1 byte
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samples_read++;
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}
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}
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if (bytes_per_sample == 1) {
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sample = audio_sample - 0x80;
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audio_sample_s16 = sample * 256;
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} else {
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audio_sample_s16 = (int16_t)audio_sample;
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sample = audio_sample_s16 / 256;
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}
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// Output to speaker too
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if (!tone_key_enabled) {
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uint32_t imod32 = i & (AUDIO_OUTPUT_BUFFER_SIZE - 1);
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audio_data[imod32] = audio_sample_s16;
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if (imod32 == (AUDIO_OUTPUT_BUFFER_SIZE - 1))
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audio_output.write_unprocessed(audio_buffer);
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}
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sample = tone_gen.process(sample);
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// FM
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delta = sample * fm_delta;
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phase += delta;
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sphase = phase + (64 << 24);
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re = sine_table_i8[(sphase & 0xFF000000U) >> 24];
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im = sine_table_i8[(phase & 0xFF000000U) >> 24];
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buffer.p[i] = {(int8_t)re, (int8_t)im};
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}
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progress_samples += buffer.count;
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if (progress_samples >= progress_interval_samples) {
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progress_samples -= progress_interval_samples;
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txprogress_message.progress = samples_read; // Inform UI about progress
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txprogress_message.done = false;
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shared_memory.application_queue.push(txprogress_message);
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}
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}
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void AudioTXProcessor::on_message(const Message* const message) {
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switch (message->id) {
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case Message::ID::AudioTXConfig:
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audio_config(*reinterpret_cast<const AudioTXConfigMessage*>(message));
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break;
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case Message::ID::ReplayConfig:
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configured = false;
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samples_read = 0;
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replay_config(*reinterpret_cast<const ReplayConfigMessage*>(message));
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break;
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case Message::ID::SampleRateConfig:
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sample_rate_config(*reinterpret_cast<const SampleRateConfigMessage*>(message));
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break;
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case Message::ID::FIFOData:
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configured = true;
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break;
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default:
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break;
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}
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}
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void AudioTXProcessor::audio_config(const AudioTXConfigMessage& message) {
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fm_delta = message.deviation_hz * (0xFFFFFFULL / baseband_fs);
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tone_gen.configure(message.tone_key_delta, message.tone_key_mix_weight);
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progress_interval_samples = message.divider;
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resample_acc = 0;
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bytes_per_sample = message.bits_per_sample / 8;
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audio_output.configure(false);
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tone_key_enabled = (message.tone_key_delta != 0);
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audio::dma::shrink_tx_buffer(!tone_key_enabled);
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}
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void AudioTXProcessor::replay_config(const ReplayConfigMessage& message) {
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if (message.config) {
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stream = std::make_unique<StreamOutput>(message.config);
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// Tell application that the buffers and FIFO pointers are ready, prefill
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shared_memory.application_queue.push(sig_message);
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} else {
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stream.reset();
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}
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}
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void AudioTXProcessor::sample_rate_config(const SampleRateConfigMessage& message) {
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resample_inc = (((uint64_t)message.sample_rate) << 16) / baseband_fs; // 16.16 fixed point message.sample_rate
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sampling_rate = message.sample_rate;
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}
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int main() {
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EventDispatcher event_dispatcher{std::make_unique<AudioTXProcessor>()};
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event_dispatcher.run();
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return 0;
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}
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