rtaudio hello world

src/main.cpp

@@ -6,6 +6,7 @@
 #include <QQmlContext>
 
 #include "TimerComponent.hpp"
+#include "synth/AudioEngine.hpp"
 
 int main(int argc, char* argv[]) {
 
@@ -27,6 +28,10 @@ int main(int argc, char* argv[]) {
         return -1;
     }
 
+    // audio synthesizer doohickey
+    AudioEngine audioEngine = AudioEngine();
+    audioEngine.start();
+
     // execute app
     return app.exec();
 }

src/synth/AudioEngine.cpp

@@ -1,2 +1,85 @@
 
 #include "AudioEngine.hpp"
+
+#include <iostream>
+#include <cmath>
+#include <numbers>
+
+AudioEngine::AudioEngine() {
+
+    if(audioDevice_.getDeviceCount() < 1) {
+        std::cout << "No audio devices found" << std::endl;
+    }
+
+}
+
+AudioEngine::~AudioEngine() {
+    (void)stop();
+}
+
+bool AudioEngine::start() {
+
+    // initialize the audio engine
+    RtAudio::StreamParameters params;
+    params.deviceId = audioDevice_.getDefaultOutputDevice();
+    params.nChannels = channels_; // we're doing two duplicate channels for pseudo-mono
+    params.firstChannel = 0;
+
+    RtAudio::StreamOptions options;
+    options.flags = RTAUDIO_MINIMIZE_LATENCY;
+
+    RtAudioErrorType status = audioDevice_.openStream(&params, nullptr, RTAUDIO_FLOAT32, sampleRate_, &bufferFrames_, &AudioEngine::audioCallback, this, &options);
+    if(status != RTAUDIO_NO_ERROR) {
+        std::cout << "Error opening RtAudio stream" << std::endl;
+        return false;
+    } 
+    
+    status = audioDevice_.startStream();
+    if(status != RTAUDIO_NO_ERROR) {
+        std::cout << "Error starting RtAudio stream" << std::endl;
+        return false;
+    } 
+
+    // sanity check
+    std::cout << "sample rate: " << sampleRate_ << "  buffer frames: " << bufferFrames_ << std::endl;
+    return true;
+
+}
+
+bool AudioEngine::stop() {
+
+    if(audioDevice_.isStreamRunning()) audioDevice_.stopStream();
+    if(audioDevice_.isStreamOpen()) audioDevice_.closeStream();
+
+    return true;
+
+}
+
+int32_t AudioEngine::audioCallback(void* outputBuffer, void* inputBuffer, uint32_t nFrames, double, RtAudioStreamStatus status, void* userData) {
+
+    // error if the callback is late
+    if(status) std::cout << "stream underflow" << std::endl;
+
+    return static_cast<AudioEngine*>(userData)->process(static_cast<float*>(outputBuffer), static_cast<size_t>(nFrames));
+
+}
+
+int32_t AudioEngine::process(float* out, size_t nFrames) {
+
+    for(size_t i = 0; i < nFrames; i++) {
+
+        // simulate a sine wave
+        phase_ += 0.04f;
+        const float twoPi = 2.0f*std::numbers::pi_v<float>;
+        if(phase_ > twoPi) {
+            phase_ -= twoPi;
+        }
+        float outSample = std::sin(phase_) / 4.0f;
+
+        out[2*i] = outSample;
+        out[2*i+1] = outSample;
+    }
+
+    return 0;
+
+}

src/synth/AudioEngine.hpp

@@ -1,2 +1,39 @@
 
 // The audio engine handles the RtAudio implementation of outputing samples to audio. the audio callback fills a buffer of audio samples and submits it for playback
+
+#pragma once
+
+#include <RtAudio.h>
+
+#if defined(_WIN32)
+    #define AUDIO_API RtAudio::WINDOWS_WASAPI
+#else
+    #define AUDIO_API RtAudio::LINUX_ALSA
+#endif
+
+class AudioEngine {
+
+public:
+
+    AudioEngine();
+    ~AudioEngine();
+
+    bool start();
+    bool stop();
+
+private:
+
+    static int32_t audioCallback(void* outputBuffer, void* inputBuffer, uint32_t nFrames, double streamtime, RtAudioStreamStatus status, void* userData);
+
+    int32_t process(float* out, size_t nFrames);
+
+    RtAudio audioDevice_ { AUDIO_API };
+
+    // TODO: make these configurable
+    uint32_t sampleRate_ = 44100;
+    uint32_t bufferFrames_ = 512;
+    uint32_t channels_ = 2;
+
+    float phase_ = 0.0f;
+
+};