refactor(soundio): Use SampleRate type instead of double

src/engine/sidechain/enginenetworkstream.cpp

@@ -36,15 +36,15 @@ const mixxx::Logger kLogger("EngineNetworkStream");
 } // namespace
 
 EngineNetworkStream::EngineNetworkStream(int numOutputChannels,
-                                         int numInputChannels)
-    : m_pInputFifo(nullptr),
-      m_numOutputChannels(numOutputChannels),
-      m_numInputChannels(numInputChannels),
-      m_sampleRate(0),
-      m_inputStreamStartTimeUs(-1),
-      m_inputStreamFramesWritten(0),
-      m_inputStreamFramesRead(0),
-      m_outputWorkers(BROADCAST_MAX_CONNECTIONS) {
+        int numInputChannels)
+        : m_pInputFifo(nullptr),
+          m_numOutputChannels(numOutputChannels),
+          m_numInputChannels(numInputChannels),
+          m_sampleRate(),
+          m_inputStreamStartTimeUs(-1),
+          m_inputStreamFramesWritten(0),
+          m_inputStreamFramesRead(0),
+          m_outputWorkers(BROADCAST_MAX_CONNECTIONS) {
     if (numInputChannels) {
         m_pInputFifo = new FIFO<CSAMPLE>(numInputChannels * kBufferFrames);
     }
@@ -73,7 +73,7 @@ EngineNetworkStream::~EngineNetworkStream() {
     delete m_pInputFifo;
 }
 
-void EngineNetworkStream::startStream(double sampleRate) {
+void EngineNetworkStream::startStream(mixxx::audio::SampleRate sampleRate) {
     m_sampleRate = sampleRate;
     m_inputStreamStartTimeUs = getNetworkTimeUs();
     m_inputStreamFramesWritten = 0;
@@ -122,7 +122,7 @@ void EngineNetworkStream::read(CSAMPLE* buffer, int frames) {
 
 qint64 EngineNetworkStream::getInputStreamTimeFrames() {
     return static_cast<qint64>(static_cast<double>(getInputStreamTimeUs()) *
-            m_sampleRate / 1000000.0);
+            m_sampleRate.toDouble() / 1000000.0);
 }
 
 qint64 EngineNetworkStream::getInputStreamTimeUs() {

src/engine/sidechain/enginenetworkstream.h

@@ -1,19 +1,20 @@
 #pragma once
 
-#include <engine/sidechain/networkoutputstreamworker.h>
-#include <engine/sidechain/networkinputstreamworker.h>
 #include <QVector>
 
-#include "util/types.h"
+#include "audio/types.h"
+#include "engine/sidechain/networkinputstreamworker.h"
+#include "engine/sidechain/networkoutputstreamworker.h"
 #include "util/fifo.h"
+#include "util/types.h"
 
 class EngineNetworkStream {
   public:
     EngineNetworkStream(int numOutputChannels,
             int numInputChannels);
     virtual ~EngineNetworkStream();
 
-    void startStream(double sampleRate);
+    void startStream(mixxx::audio::SampleRate sampleRate);
     void stopStream();
 
     int getReadExpected();
@@ -47,7 +48,7 @@ class EngineNetworkStream {
     FIFO<CSAMPLE>* m_pInputFifo;
     int m_numOutputChannels;
     int m_numInputChannels;
-    double m_sampleRate;
+    mixxx::audio::SampleRate m_sampleRate;
     qint64 m_inputStreamStartTimeUs;
     qint64 m_inputStreamFramesWritten;
     qint64 m_inputStreamFramesRead;

src/engine/sidechain/networkoutputstreamworker.cpp

@@ -9,15 +9,15 @@ const mixxx::Logger kLogger("NetworkStreamWorker");
 } // namespace
 
 NetworkOutputStreamWorker::NetworkOutputStreamWorker()
-    : m_sampleRate(0),
-      m_numOutputChannels(0),
-      m_workerState(NETWORKSTREAMWORKER_STATE_NEW),
-      m_functionCode(0),
-      m_runCount(0),
-      m_streamStartTimeUs(-1),
-      m_streamFramesWritten(0),
-      m_writeOverflowCount(0),
-      m_outputDrift(false) {
+        : m_sampleRate(),
+          m_numOutputChannels(0),
+          m_workerState(NETWORKSTREAMWORKER_STATE_NEW),
+          m_functionCode(0),
+          m_runCount(0),
+          m_streamStartTimeUs(-1),
+          m_streamFramesWritten(0),
+          m_writeOverflowCount(0),
+          m_outputDrift(false) {
 }
 
 void NetworkOutputStreamWorker::outputAvailable() {
@@ -31,16 +31,17 @@ QSharedPointer<FIFO<CSAMPLE>> NetworkOutputStreamWorker::getOutputFifo() {
     return QSharedPointer<FIFO<CSAMPLE>>();
 }
 
-void NetworkOutputStreamWorker::startStream(double samplerate, int numOutputChannels) {
-    m_sampleRate = samplerate;
+void NetworkOutputStreamWorker::startStream(
+        mixxx::audio::SampleRate sampleRate, int numOutputChannels) {
+    m_sampleRate = sampleRate;
     m_numOutputChannels = numOutputChannels;
 
     m_streamStartTimeUs = EngineNetworkStream::getNetworkTimeUs();
     m_streamFramesWritten = 0;
 }
 
 void NetworkOutputStreamWorker::stopStream() {
-    m_sampleRate = 0;
+    m_sampleRate = mixxx::audio::SampleRate();
     m_numOutputChannels = 0;
 
     m_streamStartTimeUs = -1;
@@ -51,7 +52,8 @@ bool NetworkOutputStreamWorker::threadWaiting() {
 }
 
 qint64 NetworkOutputStreamWorker::getStreamTimeFrames() {
-    return static_cast<qint64>(static_cast<double>(getStreamTimeUs()) * m_sampleRate / 1000000.0);
+    return static_cast<qint64>(static_cast<double>(getStreamTimeUs()) *
+            m_sampleRate.toDouble() / 1000000.0);
 }
 
 qint64 NetworkOutputStreamWorker::getStreamTimeUs() {

src/engine/sidechain/networkoutputstreamworker.h

@@ -2,8 +2,9 @@
 
 #include <QSharedPointer>
 
-#include "util/types.h"
+#include "audio/types.h"
 #include "util/fifo.h"
+#include "util/types.h"
 
 /*
  * States:
@@ -52,7 +53,7 @@ class NetworkOutputStreamWorker {
     virtual void setOutputFifo(QSharedPointer<FIFO<CSAMPLE>> pOutputFifo);
     virtual QSharedPointer<FIFO<CSAMPLE>> getOutputFifo();
 
-    void startStream(double samplerate, int numOutputChannels);
+    void startStream(mixxx::audio::SampleRate sampleRate, int numOutputChannels);
     void stopStream();
 
     virtual bool threadWaiting();
@@ -83,17 +84,17 @@ class NetworkOutputStreamWorker {
     void incRunCount();
 
 private:
-    double m_sampleRate;
-    int m_numOutputChannels;
+  mixxx::audio::SampleRate m_sampleRate;
+  int m_numOutputChannels;
 
-    int m_workerState;
-    int m_functionCode;
-    int m_runCount;
+  int m_workerState;
+  int m_functionCode;
+  int m_runCount;
 
-    qint64 m_streamStartTimeUs;
-    qint64 m_streamFramesWritten;
-    int m_writeOverflowCount;
-    bool m_outputDrift;
+  qint64 m_streamStartTimeUs;
+  qint64 m_streamFramesWritten;
+  int m_writeOverflowCount;
+  bool m_outputDrift;
 };
 
 typedef QSharedPointer<NetworkOutputStreamWorker> NetworkOutputStreamWorkerPtr;

src/soundio/sounddevice.cpp

@@ -9,13 +9,16 @@
 #include "util/defs.h"
 #include "util/sample.h"
 
+constexpr mixxx::audio::SampleRate SoundDevice::kFallbackSampleRate =
+        mixxx::audio::SampleRate(44100);
+
 SoundDevice::SoundDevice(UserSettingsPointer config, SoundManager* sm)
         : m_pConfig(config),
           m_pSoundManager(sm),
           m_strDisplayName("Unknown Soundcard"),
           m_iNumOutputChannels(2),
           m_iNumInputChannels(2),
-          m_dSampleRate(44100.0),
+          m_sampleRate(kFallbackSampleRate),
           m_hostAPI("Unknown API"),
           m_configFramesPerBuffer(0) {
 }
@@ -28,12 +31,8 @@ int SoundDevice::getNumOutputChannels() const {
     return m_iNumOutputChannels;
 }
 
-void SoundDevice::setSampleRate(double sampleRate) {
-    if (sampleRate <= 0.0) {
-        // this is the default value used elsewhere in this file
-        sampleRate = 44100.0;
-    }
-    m_dSampleRate = sampleRate;
+void SoundDevice::setSampleRate(mixxx::audio::SampleRate sampleRate) {
+    m_sampleRate = sampleRate.isValid() ? sampleRate : kFallbackSampleRate;
 }
 
 void SoundDevice::setConfigFramesPerBuffer(unsigned int framesPerBuffer) {

src/soundio/sounddevice.h

@@ -3,6 +3,7 @@
 #include <QList>
 #include <QString>
 
+#include "audio/types.h"
 #include "preferences/usersettings.h"
 #include "soundio/sounddevicestatus.h"
 #include "soundio/soundmanagerutil.h"
@@ -31,15 +32,15 @@ class SoundDevice {
     inline const QString& getHostAPI() const {
         return m_hostAPI;
     }
-    void setSampleRate(double sampleRate);
+    void setSampleRate(mixxx::audio::SampleRate sampleRate);
     void setConfigFramesPerBuffer(unsigned int framesPerBuffer);
     virtual SoundDeviceStatus open(bool isClkRefDevice, int syncBuffers) = 0;
     virtual bool isOpen() const = 0;
     virtual SoundDeviceStatus close() = 0;
     virtual void readProcess(SINT framesPerBuffer) = 0;
     virtual void writeProcess(SINT framesPerBuffer) = 0;
     virtual QString getError() const = 0;
-    virtual unsigned int getDefaultSampleRate() const = 0;
+    virtual mixxx::audio::SampleRate getDefaultSampleRate() const = 0;
     int getNumOutputChannels() const;
     int getNumInputChannels() const;
     SoundDeviceStatus addOutput(const AudioOutputBuffer& out);
@@ -57,6 +58,8 @@ class SoundDevice {
     bool operator==(const QString &other) const;
 
   protected:
+    static const mixxx::audio::SampleRate kFallbackSampleRate;
+
     void composeOutputBuffer(CSAMPLE* outputBuffer,
                              const SINT iFramesPerBuffer,
                              const SINT readOffset,
@@ -81,7 +84,7 @@ class SoundDevice {
     // The number of input channels that the soundcard has
     int m_iNumInputChannels;
     // The current samplerate for the sound device.
-    double m_dSampleRate;
+    mixxx::audio::SampleRate m_sampleRate;
     // The name of the audio API used by this device.
     QString m_hostAPI;
     // The **configured** number of frames per buffer. We'll tell PortAudio we

src/soundio/sounddevicenetwork.cpp

@@ -26,6 +26,8 @@ constexpr int kNetworkLatencyFrames = 8192; // 185 ms @ 44100 Hz
 // Which results in case of ogg in a dynamic latency from 0.14 ms to to 185 ms
 // Now we have switched to a fixed latency of 8192 frames (stereo samples) =
 // which is 185 @ 44100 ms and twice the maximum of the max mixxx audio buffer
+//
+constexpr auto kDefaultSampleRate = mixxx::audio::SampleRate(44100);
 
 const mixxx::Logger kLogger("SoundDeviceNetwork");
 } // namespace
@@ -43,7 +45,7 @@ SoundDeviceNetwork::SoundDeviceNetwork(
           m_targetTime(0) {
     // Setting parent class members:
     m_hostAPI = "Network stream";
-    m_dSampleRate = 44100.0;
+    m_sampleRate = SoundDevice::kFallbackSampleRate;
     m_deviceId.name = kNetworkDeviceInternalName;
     m_strDisplayName = QObject::tr("Network stream");
     m_iNumInputChannels = pNetworkStream->getNumInputChannels();
@@ -58,13 +60,13 @@ SoundDeviceStatus SoundDeviceNetwork::open(bool isClkRefDevice, int syncBuffers)
     kLogger.debug() << "open:" << m_deviceId.name;
 
     // Sample rate
-    if (m_dSampleRate <= 0) {
-        m_dSampleRate = 44100.0;
+    if (!m_sampleRate.isValid()) {
+        m_sampleRate = SoundDevice::kFallbackSampleRate;
     }
 
     const SINT framesPerBuffer = m_configFramesPerBuffer;
     const auto requestedBufferTime = mixxx::Duration::fromSeconds(
-            framesPerBuffer / m_dSampleRate);
+            framesPerBuffer / m_sampleRate.toDouble());
 
     // Feed the network device buffer directly from the
     // clock reference device callback
@@ -78,18 +80,18 @@ SoundDeviceStatus SoundDeviceNetwork::open(bool isClkRefDevice, int syncBuffers)
                 m_iNumInputChannels * framesPerBuffer * 2);
     }
 
-    m_pNetworkStream->startStream(m_dSampleRate);
+    m_pNetworkStream->startStream(m_sampleRate);
 
     // Create the callback Thread if requested
     if (isClkRefDevice) {
         kLogger.debug() << "Clock Reference with:" << framesPerBuffer << "frames/buffer @"
-                        << m_dSampleRate << "Hz =" << requestedBufferTime.formatMillisWithUnit();
+                        << m_sampleRate << "Hz =" << requestedBufferTime.formatMillisWithUnit();
 
         // Update the samplerate and latency ControlObjects, which allow the
         // waveform view to properly correct for the latency.
         ControlObject::set(ConfigKey("[Master]", "latency"),
                 requestedBufferTime.toDoubleMillis());
-        ControlObject::set(ConfigKey("[Master]", "samplerate"), m_dSampleRate);
+        ControlObject::set(ConfigKey("[Master]", "samplerate"), m_sampleRate);
 
         // Network stream was just started above so we have to wait until
         // we can pass one chunk.
@@ -100,7 +102,7 @@ SoundDeviceStatus SoundDeviceNetwork::open(bool isClkRefDevice, int syncBuffers)
         m_pThread->start(QThread::TimeCriticalPriority);
     } else {
         kLogger.debug() << "Maximum:" << framesPerBuffer << "frames/buffer @"
-                        << m_dSampleRate << "Hz =" << requestedBufferTime.formatMillisWithUnit();
+                        << m_sampleRate << "Hz =" << requestedBufferTime.formatMillisWithUnit();
     }
 
     return SoundDeviceStatus::Ok;
@@ -125,6 +127,10 @@ SoundDeviceStatus SoundDeviceNetwork::close() {
     return SoundDeviceStatus::Ok;
 }
 
+mixxx::audio::SampleRate SoundDeviceNetwork::getDefaultSampleRate() const {
+    return kDefaultSampleRate;
+}
+
 QString SoundDeviceNetwork::getError() const {
     return QString();
 }
@@ -501,7 +507,7 @@ void SoundDeviceNetwork::updateCallbackEntryToDacTime(SINT framesPerBuffer) {
     m_clkRefTimer.start();
     qint64 currentTime = m_pNetworkStream->getInputStreamTimeUs();
     // This deadline for the next buffer in microseconds since the Unix epoch
-    m_targetTime += static_cast<qint64>(framesPerBuffer / m_dSampleRate * 1000000);
+    m_targetTime += static_cast<qint64>(framesPerBuffer / m_sampleRate.toDouble() * 1000000);
     double callbackEntrytoDacSecs = (m_targetTime - currentTime) / 1000000.0;
     callbackEntrytoDacSecs = math_max(callbackEntrytoDacSecs, 0.0001);
     VisualPlayPosition::setCallbackEntryToDacSecs(callbackEntrytoDacSecs, m_clkRefTimer);
@@ -510,11 +516,10 @@ void SoundDeviceNetwork::updateCallbackEntryToDacTime(SINT framesPerBuffer) {
 
 void SoundDeviceNetwork::updateAudioLatencyUsage(SINT framesPerBuffer) {
     m_framesSinceAudioLatencyUsageUpdate += framesPerBuffer;
-    if (m_framesSinceAudioLatencyUsageUpdate
-            > (m_dSampleRate / CPU_USAGE_UPDATE_RATE)) {
+    if (m_framesSinceAudioLatencyUsageUpdate > (m_sampleRate.toDouble() / CPU_USAGE_UPDATE_RATE)) {
         double secInAudioCb = m_timeInAudioCallback.toDoubleSeconds();
         m_masterAudioLatencyUsage.set(secInAudioCb /
-                (m_framesSinceAudioLatencyUsageUpdate / m_dSampleRate));
+                (m_framesSinceAudioLatencyUsageUpdate / m_sampleRate.toDouble()));
         m_timeInAudioCallback = mixxx::Duration::empty();
         m_framesSinceAudioLatencyUsageUpdate = 0;
         //qDebug() << m_pMasterAudioLatencyUsage->get();

src/soundio/sounddevicenetwork.h

@@ -22,7 +22,6 @@ class ControlProxy;
 class EngineNetworkStream;
 class SoundDeviceNetworkThread;
 
-
 class SoundDeviceNetwork : public SoundDevice {
   public:
     SoundDeviceNetwork(UserSettingsPointer config,
@@ -37,9 +36,7 @@ class SoundDeviceNetwork : public SoundDevice {
     void writeProcess(SINT framesPerBuffer) override;
     QString getError() const override;
 
-    unsigned int getDefaultSampleRate() const override {
-        return 44100;
-    }
+    mixxx::audio::SampleRate getDefaultSampleRate() const override;
 
     // NOTE: This does not take a frames per buffer argument because that is
     //       always equal to the configured buffer size for network streams

src/soundio/sounddevicenotfound.h

@@ -35,7 +35,7 @@ class SoundDeviceNotFound : public SoundDevice {
         return QObject::tr("Device not found");
     };
 
-    unsigned int getDefaultSampleRate() const override {
-        return 44100;
+    mixxx::audio::SampleRate getDefaultSampleRate() const override {
+        return SoundDevice::kFallbackSampleRate;
     }
 };