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Split AudioInterface into AudioInterfacePort, avoiding multiple inheritance.

tags/v2.0.0
Andrew Belt 3 years ago
parent
1e6bc7cdf5
commit
9c7ac3392c
1 changed files with 181 additions and 163 deletions
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  1. +181
    -163
      src/core/AudioInterface.cpp

+ 181
- 163
src/core/AudioInterface.cpp View File

@@ -13,7 +13,166 @@ namespace core {




template <int NUM_AUDIO_INPUTS, int NUM_AUDIO_OUTPUTS> template <int NUM_AUDIO_INPUTS, int NUM_AUDIO_OUTPUTS>
struct AudioInterface : Module, audio::Port {
struct AudioInterfacePort : audio::Port {
Module* module;

dsp::DoubleRingBuffer<dsp::Frame<NUM_AUDIO_INPUTS>, 32768> engineInputBuffer;
dsp::DoubleRingBuffer<dsp::Frame<NUM_AUDIO_OUTPUTS>, 32768> engineOutputBuffer;

dsp::SampleRateConverter<NUM_AUDIO_INPUTS> inputSrc;
dsp::SampleRateConverter<NUM_AUDIO_OUTPUTS> outputSrc;

// Port variable caches
int deviceNumInputs = 0;
int deviceNumOutputs = 0;
float deviceSampleRate = 0.f;
int requestedEngineFrames = 0;

AudioInterfacePort(Module* module) {
this->module = module;
maxOutputs = NUM_AUDIO_INPUTS;
maxInputs = NUM_AUDIO_OUTPUTS;
inputSrc.setQuality(6);
outputSrc.setQuality(6);
}

void setPrimary() {
APP->engine->setPrimaryModule(module);
}

bool isPrimary() {
return APP->engine->getPrimaryModule() == module;
}

void processInput(const float* input, int inputStride, int frames) override {
// DEBUG("%p: new device block ____________________________", this);
// Claim primary module if there is none
if (!APP->engine->getPrimaryModule()) {
setPrimary();
}
bool isPrimaryCached = isPrimary();

// Set sample rate of engine if engine sample rate is "auto".
if (isPrimaryCached) {
APP->engine->setSuggestedSampleRate(deviceSampleRate);
}

float engineSampleRate = APP->engine->getSampleRate();
float sampleRateRatio = engineSampleRate / deviceSampleRate;

// DEBUG("%p: %d block, engineOutputBuffer still has %d", this, frames, (int) engineOutputBuffer.size());

// Consider engine buffers "too full" if they contain a bit more than the audio device's number of frames, converted to engine sample rate.
int maxEngineFrames = (int) std::ceil(frames * sampleRateRatio * 2.0) - 1;
// If the engine output buffer is too full, clear it to keep latency low. No need to clear if primary because it's always cleared below.
if (!isPrimaryCached && (int) engineOutputBuffer.size() > maxEngineFrames) {
engineOutputBuffer.clear();
// DEBUG("%p: clearing engine output", this);
}

if (deviceNumInputs > 0) {
// Always clear engine output if primary
if (isPrimaryCached) {
engineOutputBuffer.clear();
}
// Set up sample rate converter
outputSrc.setRates(deviceSampleRate, engineSampleRate);
outputSrc.setChannels(deviceNumInputs);
// Convert audio input -> engine output
dsp::Frame<NUM_AUDIO_OUTPUTS> audioInputBuffer[frames];
std::memset(audioInputBuffer, 0, sizeof(audioInputBuffer));
for (int i = 0; i < frames; i++) {
for (int j = 0; j < deviceNumInputs; j++) {
float v = input[i * inputStride + j];
audioInputBuffer[i].samples[j] = v;
}
}
int audioInputFrames = frames;
int outputFrames = engineOutputBuffer.capacity();
outputSrc.process(audioInputBuffer, &audioInputFrames, engineOutputBuffer.endData(), &outputFrames);
engineOutputBuffer.endIncr(outputFrames);
// Request exactly as many frames as we have in the engine output buffer.
requestedEngineFrames = engineOutputBuffer.size();
}
else {
// Upper bound on number of frames so that `audioOutputFrames >= frames` when processOutput() is called.
requestedEngineFrames = std::max((int) std::ceil(frames * sampleRateRatio) - (int) engineInputBuffer.size(), 0);
}
}

void processBuffer(const float* input, int inputStride, float* output, int outputStride, int frames) override {
// Step engine
if (isPrimary() && requestedEngineFrames > 0) {
// DEBUG("%p: %d block, stepping %d", this, frames, requestedEngineFrames);
APP->engine->stepBlock(requestedEngineFrames);
}
}

void processOutput(float* output, int outputStride, int frames) override {
// bool isPrimaryCached = isPrimary();
float engineSampleRate = APP->engine->getSampleRate();
float sampleRateRatio = engineSampleRate / deviceSampleRate;

if (deviceNumOutputs > 0) {
// Set up sample rate converter
inputSrc.setRates(engineSampleRate, deviceSampleRate);
inputSrc.setChannels(deviceNumOutputs);
// Convert engine input -> audio output
dsp::Frame<NUM_AUDIO_OUTPUTS> audioOutputBuffer[frames];
int inputFrames = engineInputBuffer.size();
int audioOutputFrames = frames;
inputSrc.process(engineInputBuffer.startData(), &inputFrames, audioOutputBuffer, &audioOutputFrames);
engineInputBuffer.startIncr(inputFrames);
// Copy the audio output buffer
for (int i = 0; i < audioOutputFrames; i++) {
for (int j = 0; j < deviceNumOutputs; j++) {
float v = audioOutputBuffer[i].samples[j];
v = clamp(v, -1.f, 1.f);
output[i * outputStride + j] = v;
}
}
// Fill the rest of the audio output buffer with zeros
for (int i = audioOutputFrames; i < frames; i++) {
for (int j = 0; j < deviceNumOutputs; j++) {
output[i * outputStride + j] = 0.f;
}
}
}

// DEBUG("%p: %d block, engineInputBuffer left %d", this, frames, (int) engineInputBuffer.size());

// If the engine input buffer is too full, clear it to keep latency low.
int maxEngineFrames = (int) std::ceil(frames * sampleRateRatio * 2.0) - 1;
if ((int) engineInputBuffer.size() > maxEngineFrames) {
engineInputBuffer.clear();
// DEBUG("%p: clearing engine input", this);
}

// DEBUG("%p %s:\tframes %d requestedEngineFrames %d\toutputBuffer %d engineInputBuffer %d\t", this, isPrimaryCached ? "primary" : "secondary", frames, requestedEngineFrames, engineOutputBuffer.size(), engineInputBuffer.size());
}

void onStartStream() override {
deviceNumInputs = std::min(getNumInputs(), NUM_AUDIO_OUTPUTS);
deviceNumOutputs = std::min(getNumOutputs(), NUM_AUDIO_INPUTS);
deviceSampleRate = getSampleRate();
engineInputBuffer.clear();
engineOutputBuffer.clear();
// DEBUG("onStartStream %d %d %f", deviceNumInputs, deviceNumOutputs, deviceSampleRate);
}

void onStopStream() override {
deviceNumInputs = 0;
deviceNumOutputs = 0;
deviceSampleRate = 0.f;
engineInputBuffer.clear();
engineOutputBuffer.clear();
// DEBUG("onStopStream");
}
};


template <int NUM_AUDIO_INPUTS, int NUM_AUDIO_OUTPUTS>
struct AudioInterface : Module {
static constexpr int NUM_INPUT_LIGHTS = (NUM_AUDIO_INPUTS > 2) ? (NUM_AUDIO_INPUTS / 2) : 0; static constexpr int NUM_INPUT_LIGHTS = (NUM_AUDIO_INPUTS > 2) ? (NUM_AUDIO_INPUTS / 2) : 0;
static constexpr int NUM_OUTPUT_LIGHTS = (NUM_AUDIO_OUTPUTS > 2) ? (NUM_AUDIO_OUTPUTS / 2) : 0; static constexpr int NUM_OUTPUT_LIGHTS = (NUM_AUDIO_OUTPUTS > 2) ? (NUM_AUDIO_OUTPUTS / 2) : 0;


@@ -36,28 +195,18 @@ struct AudioInterface : Module, audio::Port {
NUM_LIGHTS NUM_LIGHTS
}; };


AudioInterfacePort<NUM_AUDIO_INPUTS, NUM_AUDIO_OUTPUTS> port;

dsp::RCFilter dcFilters[NUM_AUDIO_INPUTS]; dsp::RCFilter dcFilters[NUM_AUDIO_INPUTS];
bool dcFilterEnabled = false; bool dcFilterEnabled = false;


dsp::DoubleRingBuffer<dsp::Frame<NUM_AUDIO_INPUTS>, 32768> engineInputBuffer;
dsp::DoubleRingBuffer<dsp::Frame<NUM_AUDIO_OUTPUTS>, 32768> engineOutputBuffer;

dsp::SampleRateConverter<NUM_AUDIO_INPUTS> inputSrc;
dsp::SampleRateConverter<NUM_AUDIO_OUTPUTS> outputSrc;

dsp::ClockDivider lightDivider; dsp::ClockDivider lightDivider;
// For each pair of inputs/outputs // For each pair of inputs/outputs
float inputClipTimers[(NUM_AUDIO_INPUTS > 0) ? NUM_INPUT_LIGHTS : 0] = {}; float inputClipTimers[(NUM_AUDIO_INPUTS > 0) ? NUM_INPUT_LIGHTS : 0] = {};
float outputClipTimers[(NUM_AUDIO_INPUTS > 0) ? NUM_OUTPUT_LIGHTS : 0] = {}; float outputClipTimers[(NUM_AUDIO_INPUTS > 0) ? NUM_OUTPUT_LIGHTS : 0] = {};
dsp::VuMeter2 vuMeter[(NUM_AUDIO_INPUTS == 2) ? 2 : 0]; dsp::VuMeter2 vuMeter[(NUM_AUDIO_INPUTS == 2) ? 2 : 0];


// Port variable caches
int deviceNumInputs = 0;
int deviceNumOutputs = 0;
float deviceSampleRate = 0.f;
int requestedEngineFrames = 0;

AudioInterface() {
AudioInterface() : port(this) {
config(NUM_PARAMS, NUM_INPUTS, NUM_OUTPUTS, NUM_LIGHTS); config(NUM_PARAMS, NUM_INPUTS, NUM_OUTPUTS, NUM_LIGHTS);
if (NUM_AUDIO_INPUTS == 2) if (NUM_AUDIO_INPUTS == 2)
configParam(GAIN_PARAM, 0.f, 2.f, 1.f, "Level", " dB", -10, 40); configParam(GAIN_PARAM, 0.f, 2.f, 1.f, "Level", " dB", -10, 40);
@@ -71,26 +220,22 @@ struct AudioInterface : Module, audio::Port {
configLight(OUTPUT_LIGHTS + 2 * i, string::f("Device input %d/%d status", 2 * i + 1, 2 * i + 2)); configLight(OUTPUT_LIGHTS + 2 * i, string::f("Device input %d/%d status", 2 * i + 1, 2 * i + 2));


lightDivider.setDivision(512); lightDivider.setDivision(512);
maxOutputs = NUM_AUDIO_INPUTS;
maxInputs = NUM_AUDIO_OUTPUTS;
inputSrc.setQuality(6);
outputSrc.setQuality(6);


float sampleTime = APP->engine->getSampleTime(); float sampleTime = APP->engine->getSampleTime();
for (int i = 0; i < NUM_AUDIO_INPUTS; i++) { for (int i = 0; i < NUM_AUDIO_INPUTS; i++) {
dcFilters[i].setCutoffFreq(10.f * sampleTime); dcFilters[i].setCutoffFreq(10.f * sampleTime);
} }


reset();
onReset();
} }


~AudioInterface() { ~AudioInterface() {
// Close stream here before destructing AudioInterfacePort, so processBuffer() etc are not called on another thread while destructing. // Close stream here before destructing AudioInterfacePort, so processBuffer() etc are not called on another thread while destructing.
setDriverId(-1);
port.setDriverId(-1);
} }


void onReset() override { void onReset() override {
setDriverId(-1);
port.setDriverId(-1);


if (NUM_AUDIO_INPUTS == 2) if (NUM_AUDIO_INPUTS == 2)
dcFilterEnabled = true; dcFilterEnabled = true;
@@ -99,8 +244,8 @@ struct AudioInterface : Module, audio::Port {
} }


void onSampleRateChange(const SampleRateChangeEvent& e) override { void onSampleRateChange(const SampleRateChangeEvent& e) override {
engineInputBuffer.clear();
engineOutputBuffer.clear();
port.engineInputBuffer.clear();
port.engineOutputBuffer.clear();


for (int i = 0; i < NUM_AUDIO_INPUTS; i++) { for (int i = 0; i < NUM_AUDIO_INPUTS; i++) {
dcFilters[i].setCutoffFreq(10.f * e.sampleTime); dcFilters[i].setCutoffFreq(10.f * e.sampleTime);
@@ -111,9 +256,9 @@ struct AudioInterface : Module, audio::Port {
const float clipTime = 0.25f; const float clipTime = 0.25f;


// Push inputs to buffer // Push inputs to buffer
if (deviceNumOutputs > 0) {
if (port.deviceNumOutputs > 0) {
dsp::Frame<NUM_AUDIO_INPUTS> inputFrame = {}; dsp::Frame<NUM_AUDIO_INPUTS> inputFrame = {};
for (int i = 0; i < deviceNumOutputs; i++) {
for (int i = 0; i < port.deviceNumOutputs; i++) {
// Get input // Get input
float v = 0.f; float v = 0.f;
if (inputs[AUDIO_INPUTS + i].isConnected()) if (inputs[AUDIO_INPUTS + i].isConnected())
@@ -144,8 +289,8 @@ struct AudioInterface : Module, audio::Port {
} }
} }


if (!engineInputBuffer.full()) {
engineInputBuffer.push(inputFrame);
if (!port.engineInputBuffer.full()) {
port.engineInputBuffer.push(inputFrame);
} }


// Audio-2: VU meter process // Audio-2: VU meter process
@@ -165,8 +310,8 @@ struct AudioInterface : Module, audio::Port {
} }


// Pull outputs from buffer // Pull outputs from buffer
if (!engineOutputBuffer.empty()) {
dsp::Frame<NUM_AUDIO_OUTPUTS> outputFrame = engineOutputBuffer.shift();
if (!port.engineOutputBuffer.empty()) {
dsp::Frame<NUM_AUDIO_OUTPUTS> outputFrame = port.engineOutputBuffer.shift();
for (int i = 0; i < NUM_AUDIO_OUTPUTS; i++) { for (int i = 0; i < NUM_AUDIO_OUTPUTS; i++) {
float v = outputFrame.samples[i]; float v = outputFrame.samples[i];
outputs[AUDIO_OUTPUTS + i].setVoltage(10.f * v); outputs[AUDIO_OUTPUTS + i].setVoltage(10.f * v);
@@ -203,7 +348,7 @@ struct AudioInterface : Module, audio::Port {
else { else {
// Turn on light if at least one port is enabled in the nearby pair. // Turn on light if at least one port is enabled in the nearby pair.
for (int i = 0; i < NUM_AUDIO_INPUTS / 2; i++) { for (int i = 0; i < NUM_AUDIO_INPUTS / 2; i++) {
bool active = deviceNumOutputs >= 2 * i + 1;
bool active = port.deviceNumOutputs >= 2 * i + 1;
bool clip = inputClipTimers[i] > 0.f; bool clip = inputClipTimers[i] > 0.f;
if (clip) if (clip)
inputClipTimers[i] -= lightTime; inputClipTimers[i] -= lightTime;
@@ -211,7 +356,7 @@ struct AudioInterface : Module, audio::Port {
lights[INPUT_LIGHTS + i * 2 + 1].setBrightness(active && clip); lights[INPUT_LIGHTS + i * 2 + 1].setBrightness(active && clip);
} }
for (int i = 0; i < NUM_AUDIO_OUTPUTS / 2; i++) { for (int i = 0; i < NUM_AUDIO_OUTPUTS / 2; i++) {
bool active = deviceNumInputs >= 2 * i + 1;
bool active = port.deviceNumInputs >= 2 * i + 1;
bool clip = outputClipTimers[i] > 0.f; bool clip = outputClipTimers[i] > 0.f;
if (clip) if (clip)
outputClipTimers[i] -= lightTime; outputClipTimers[i] -= lightTime;
@@ -224,7 +369,7 @@ struct AudioInterface : Module, audio::Port {


json_t* dataToJson() override { json_t* dataToJson() override {
json_t* rootJ = json_object(); json_t* rootJ = json_object();
json_object_set_new(rootJ, "audio", audio::Port::toJson());
json_object_set_new(rootJ, "audio", port.toJson());


if (isPrimary()) if (isPrimary())
json_object_set_new(rootJ, "primary", json_boolean(true)); json_object_set_new(rootJ, "primary", json_boolean(true));
@@ -237,7 +382,7 @@ struct AudioInterface : Module, audio::Port {
void dataFromJson(json_t* rootJ) override { void dataFromJson(json_t* rootJ) override {
json_t* audioJ = json_object_get(rootJ, "audio"); json_t* audioJ = json_object_get(rootJ, "audio");
if (audioJ) if (audioJ)
audio::Port::fromJson(audioJ);
port.fromJson(audioJ);


// For not, don't deserialize primary module state. // For not, don't deserialize primary module state.
// json_t* primaryJ = json_object_get(rootJ, "primary"); // json_t* primaryJ = json_object_get(rootJ, "primary");
@@ -258,133 +403,6 @@ struct AudioInterface : Module, audio::Port {
bool isPrimary() { bool isPrimary() {
return APP->engine->getPrimaryModule() == this; return APP->engine->getPrimaryModule() == this;
} }

// audio::Port

void processInput(const float* input, int inputStride, int frames) override {
// DEBUG("%p: new device block ____________________________", this);
// Claim primary module if there is none
if (!APP->engine->getPrimaryModule()) {
setPrimary();
}
bool isPrimaryCached = isPrimary();

// Set sample rate of engine if engine sample rate is "auto".
if (isPrimaryCached) {
APP->engine->setSuggestedSampleRate(deviceSampleRate);
}

float engineSampleRate = APP->engine->getSampleRate();
float sampleRateRatio = engineSampleRate / deviceSampleRate;

// DEBUG("%p: %d block, engineOutputBuffer still has %d", this, frames, (int) engineOutputBuffer.size());

// Consider engine buffers "too full" if they contain a bit more than the audio device's number of frames, converted to engine sample rate.
int maxEngineFrames = (int) std::ceil(frames * sampleRateRatio * 2.0) - 1;
// If the engine output buffer is too full, clear it to keep latency low. No need to clear if primary because it's always cleared below.
if (!isPrimaryCached && (int) engineOutputBuffer.size() > maxEngineFrames) {
engineOutputBuffer.clear();
// DEBUG("%p: clearing engine output", this);
}

if (deviceNumInputs > 0) {
// Always clear engine output if primary
if (isPrimaryCached) {
engineOutputBuffer.clear();
}
// Set up sample rate converter
outputSrc.setRates(deviceSampleRate, engineSampleRate);
outputSrc.setChannels(deviceNumInputs);
// Convert audio input -> engine output
dsp::Frame<NUM_AUDIO_OUTPUTS> audioInputBuffer[frames];
std::memset(audioInputBuffer, 0, sizeof(audioInputBuffer));
for (int i = 0; i < frames; i++) {
for (int j = 0; j < deviceNumInputs; j++) {
float v = input[i * inputStride + j];
audioInputBuffer[i].samples[j] = v;
}
}
int audioInputFrames = frames;
int outputFrames = engineOutputBuffer.capacity();
outputSrc.process(audioInputBuffer, &audioInputFrames, engineOutputBuffer.endData(), &outputFrames);
engineOutputBuffer.endIncr(outputFrames);
// Request exactly as many frames as we have in the engine output buffer.
requestedEngineFrames = engineOutputBuffer.size();
}
else {
// Upper bound on number of frames so that `audioOutputFrames >= frames` when processOutput() is called.
requestedEngineFrames = std::max((int) std::ceil(frames * sampleRateRatio) - (int) engineInputBuffer.size(), 0);
}
}

void processBuffer(const float* input, int inputStride, float* output, int outputStride, int frames) override {
// Step engine
if (isPrimary() && requestedEngineFrames > 0) {
// DEBUG("%p: %d block, stepping %d", this, frames, requestedEngineFrames);
APP->engine->stepBlock(requestedEngineFrames);
}
}

void processOutput(float* output, int outputStride, int frames) override {
// bool isPrimaryCached = isPrimary();
float engineSampleRate = APP->engine->getSampleRate();
float sampleRateRatio = engineSampleRate / deviceSampleRate;

if (deviceNumOutputs > 0) {
// Set up sample rate converter
inputSrc.setRates(engineSampleRate, deviceSampleRate);
inputSrc.setChannels(deviceNumOutputs);
// Convert engine input -> audio output
dsp::Frame<NUM_AUDIO_OUTPUTS> audioOutputBuffer[frames];
int inputFrames = engineInputBuffer.size();
int audioOutputFrames = frames;
inputSrc.process(engineInputBuffer.startData(), &inputFrames, audioOutputBuffer, &audioOutputFrames);
engineInputBuffer.startIncr(inputFrames);
// Copy the audio output buffer
for (int i = 0; i < audioOutputFrames; i++) {
for (int j = 0; j < deviceNumOutputs; j++) {
float v = audioOutputBuffer[i].samples[j];
v = clamp(v, -1.f, 1.f);
output[i * outputStride + j] = v;
}
}
// Fill the rest of the audio output buffer with zeros
for (int i = audioOutputFrames; i < frames; i++) {
for (int j = 0; j < deviceNumOutputs; j++) {
output[i * outputStride + j] = 0.f;
}
}
}

// DEBUG("%p: %d block, engineInputBuffer left %d", this, frames, (int) engineInputBuffer.size());

// If the engine input buffer is too full, clear it to keep latency low.
int maxEngineFrames = (int) std::ceil(frames * sampleRateRatio * 2.0) - 1;
if ((int) engineInputBuffer.size() > maxEngineFrames) {
engineInputBuffer.clear();
// DEBUG("%p: clearing engine input", this);
}

// DEBUG("%p %s:\tframes %d requestedEngineFrames %d\toutputBuffer %d engineInputBuffer %d\t", this, isPrimaryCached ? "primary" : "secondary", frames, requestedEngineFrames, engineOutputBuffer.size(), engineInputBuffer.size());
}

void onStartStream() override {
deviceNumInputs = std::min(getNumInputs(), NUM_AUDIO_OUTPUTS);
deviceNumOutputs = std::min(getNumOutputs(), NUM_AUDIO_INPUTS);
deviceSampleRate = getSampleRate();
engineInputBuffer.clear();
engineOutputBuffer.clear();
// DEBUG("onStartStream %d %d %f", deviceNumInputs, deviceNumOutputs, deviceSampleRate);
}

void onStopStream() override {
deviceNumInputs = 0;
deviceNumOutputs = 0;
deviceSampleRate = 0.f;
engineInputBuffer.clear();
engineOutputBuffer.clear();
// DEBUG("onStopStream");
}
}; };




@@ -432,7 +450,7 @@ struct AudioInterfaceWidget : ModuleWidget {


AudioWidget* audioWidget = createWidget<AudioWidget>(mm2px(Vec(3.2122073, 14.837339))); AudioWidget* audioWidget = createWidget<AudioWidget>(mm2px(Vec(3.2122073, 14.837339)));
audioWidget->box.size = mm2px(Vec(44, 28)); audioWidget->box.size = mm2px(Vec(44, 28));
audioWidget->setAudioPort(module);
audioWidget->setAudioPort(module ? &module->port : NULL);
addChild(audioWidget); addChild(audioWidget);
} }
else if (NUM_AUDIO_INPUTS == 16 && NUM_AUDIO_OUTPUTS == 16) { else if (NUM_AUDIO_INPUTS == 16 && NUM_AUDIO_OUTPUTS == 16) {
@@ -496,7 +514,7 @@ struct AudioInterfaceWidget : ModuleWidget {


AudioWidget* audioWidget = createWidget<AudioWidget>(mm2px(Vec(2.57, 14.839))); AudioWidget* audioWidget = createWidget<AudioWidget>(mm2px(Vec(2.57, 14.839)));
audioWidget->box.size = mm2px(Vec(91.382, 28.0)); audioWidget->box.size = mm2px(Vec(91.382, 28.0));
audioWidget->setAudioPort(module);
audioWidget->setAudioPort(module ? &module->port : NULL);
addChild(audioWidget); addChild(audioWidget);
} }
else if (NUM_AUDIO_INPUTS == 2 && NUM_AUDIO_OUTPUTS == 2) { else if (NUM_AUDIO_INPUTS == 2 && NUM_AUDIO_OUTPUTS == 2) {
@@ -530,7 +548,7 @@ struct AudioInterfaceWidget : ModuleWidget {


AudioDeviceWidget* audioWidget = createWidget<AudioDeviceWidget>(mm2px(Vec(2.135, 14.259))); AudioDeviceWidget* audioWidget = createWidget<AudioDeviceWidget>(mm2px(Vec(2.135, 14.259)));
audioWidget->box.size = mm2px(Vec(21.128, 6.725)); audioWidget->box.size = mm2px(Vec(21.128, 6.725));
audioWidget->setAudioPort(module);
audioWidget->setAudioPort(module ? &module->port : NULL);
// Adjust deviceChoice position // Adjust deviceChoice position
audioWidget->deviceChoice->textOffset = Vec(6, 14); audioWidget->deviceChoice->textOffset = Vec(6, 14);
addChild(audioWidget); addChild(audioWidget);


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