Merge pull request #11121 from daschuer/jack-api-check-rebase

JACK buffer size fix.

src/engine/enginebuffer.cpp

@@ -560,7 +560,7 @@ void EngineBuffer::ejectTrack() {
     m_pTrackLoaded->forceSet(0);
     m_pTrackSamples->set(0);
     m_pTrackSampleRate->set(0);
-    m_visualPlayPos->set(0.0, 0.0, 0.0, 0.0, 0.0);
+    m_visualPlayPos->set(0.0, 0.0, 0.0, 0.0, 0.0, 0.0);
     TrackPointer pTrack = m_pCurrentTrack;
     m_pCurrentTrack.reset();
     m_playButton->set(0.0);
@@ -1298,10 +1298,13 @@ void EngineBuffer::updateIndicators(double speed, int iBufferSize) {
 
     // Update visual control object, this needs to be done more often than the
     // playpos slider
-    m_visualPlayPos->set(fFractionalPlaypos, speed * m_baserate_old,
+    m_visualPlayPos->set(
+            fFractionalPlaypos,
+            speed * m_baserate_old,
             (double)iBufferSize / m_trackSamplesOld,
             fractionalPlayposFromAbsolute(m_dSlipPosition),
-            tempoTrackSeconds);
+            tempoTrackSeconds,
+            iBufferSize / kSamplesPerFrame / static_cast<double>(m_iSampleRate) * 1000000.0);
 }
 
 void EngineBuffer::hintReader(const double dRate) {

src/engine/enginemaster.h

@@ -262,8 +262,10 @@ class EngineMaster : public QObject, public AudioSource {
     void processChannels(int iBufferSize);
 
     ChannelHandleFactoryPointer m_pChannelHandleFactory;
-    void applyMasterEffects();
-    void processHeadphones(const CSAMPLE_GAIN masterMixGainInHeadphones);
+    void applyMasterEffects(int iBufferSize);
+    void processHeadphones(
+            const CSAMPLE_GAIN masterMixGainInHeadphones,
+            int iBufferSize);
     bool sidechainMixRequired() const;
 
     EngineEffectsManager* m_pEngineEffectsManager;
@@ -284,7 +286,6 @@ class EngineMaster : public QObject, public AudioSource {
     QVarLengthArray<ChannelInfo*, kPreallocatedChannels> m_activeTalkoverChannels;
 
     unsigned int m_iSampleRate;
-    unsigned int m_iBufferSize;
 
     // Mixing buffers for each output.
     CSAMPLE* m_pOutputBusBuffers[3];
@@ -302,7 +303,6 @@ class EngineMaster : public QObject, public AudioSource {
     ControlObject* m_pHeadGain;
     ControlObject* m_pMasterSampleRate;
     ControlObject* m_pMasterLatency;
-    ControlObject* m_pMasterAudioBufferSize;
     ControlObject* m_pAudioLatencyOverloadCount;
     ControlObject* m_pNumMicsConfigured;
     ControlPotmeter* m_pAudioLatencyUsage;

src/preferences/dialog/dlgprefsound.cpp

@@ -522,10 +522,15 @@ void DlgPrefSound::apiChanged(int index) {
         sampleRateComboBox->setEnabled(false);
         latencyLabel->setEnabled(false);
         audioBufferComboBox->setEnabled(false);
+        deviceSyncComboBox->setEnabled(false);
+        engineClockComboBox->setEnabled(false);
+
     } else {
         sampleRateComboBox->setEnabled(true);
         latencyLabel->setEnabled(true);
         audioBufferComboBox->setEnabled(true);
+        deviceSyncComboBox->setEnabled(true);
+        engineClockComboBox->setEnabled(true);
     }
 }
 

src/soundio/sounddevice.cpp

@@ -17,7 +17,7 @@ SoundDevice::SoundDevice(UserSettingsPointer config, SoundManager* sm)
           m_iNumInputChannels(2),
           m_dSampleRate(44100.0),
           m_hostAPI("Unknown API"),
-          m_framesPerBuffer(0) {
+          m_configFramesPerBuffer(0) {
 }
 
 int SoundDevice::getNumInputChannels() const {
@@ -36,15 +36,15 @@ void SoundDevice::setSampleRate(double sampleRate) {
     m_dSampleRate = sampleRate;
 }
 
-void SoundDevice::setFramesPerBuffer(unsigned int framesPerBuffer) {
+void SoundDevice::setConfigFramesPerBuffer(unsigned int framesPerBuffer) {
     if (framesPerBuffer * 2 > MAX_BUFFER_LEN) {
         // framesPerBuffer * 2 because a frame will generally end up
         // being 2 samples and MAX_BUFFER_LEN is a number of samples
         // this isn't checked elsewhere, so...
         reportFatalErrorAndQuit("framesPerBuffer too big in "
                                 "SoundDevice::setFramesPerBuffer(uint)");
     }
-    m_framesPerBuffer = framesPerBuffer;
+    m_configFramesPerBuffer = framesPerBuffer;
 }
 
 SoundDeviceError SoundDevice::addOutput(const AudioOutputBuffer &out) {

src/soundio/sounddevice.h

@@ -32,12 +32,12 @@ class SoundDevice {
         return m_hostAPI;
     }
     void setSampleRate(double sampleRate);
-    void setFramesPerBuffer(unsigned int framesPerBuffer);
+    void setConfigFramesPerBuffer(unsigned int framesPerBuffer);
     virtual SoundDeviceError open(bool isClkRefDevice, int syncBuffers) = 0;
     virtual bool isOpen() const = 0;
     virtual SoundDeviceError close() = 0;
-    virtual void readProcess() = 0;
-    virtual void writeProcess() = 0;
+    virtual void readProcess(SINT framesPerBuffer) = 0;
+    virtual void writeProcess(SINT framesPerBuffer) = 0;
     virtual QString getError() const = 0;
     virtual unsigned int getDefaultSampleRate() const = 0;
     int getNumOutputChannels() const;
@@ -84,7 +84,12 @@ class SoundDevice {
     double m_dSampleRate;
     // The name of the audio API used by this device.
     QString m_hostAPI;
-    SINT m_framesPerBuffer;
+    // The **configured** number of frames per buffer. We'll tell PortAudio we
+    // want this many frames in a buffer, but PortAudio may still give us have a
+    // differently sized buffers. As such this value should only be used for
+    // configuring the audio devices. The actual runtime buffer size should be
+    // used for any computations working with audio.
+    SINT m_configFramesPerBuffer;
     QList<AudioOutputBuffer> m_audioOutputs;
     QList<AudioInputBuffer> m_audioInputs;
 };

src/soundio/sounddevicenetwork.cpp

@@ -63,23 +63,24 @@ SoundDeviceError SoundDeviceNetwork::open(bool isClkRefDevice, int syncBuffers)
         m_dSampleRate = 44100.0;
     }
 
-    qDebug() << "framesPerBuffer:" << m_framesPerBuffer;
+    const SINT framesPerBuffer = m_configFramesPerBuffer;
+    qDebug() << "framesPerBuffer:" << framesPerBuffer;
 
-    m_audioBufferTime = mixxx::Duration::fromSeconds(
-            m_framesPerBuffer / m_dSampleRate);
+    const auto requestedBufferTime = mixxx::Duration::fromSeconds(
+            framesPerBuffer / m_dSampleRate);
     qDebug() << "Requested sample rate: " << m_dSampleRate << "Hz, latency:"
-             << m_audioBufferTime;
+             << requestedBufferTime;
 
     // Feed the network device buffer directly from the
     // clock reference device callback
     // This is what should work best.
     if (m_iNumOutputChannels) {
-        m_outputFifo = std::make_unique<FIFO<CSAMPLE> >(
-                m_iNumOutputChannels * m_framesPerBuffer * 2);
+        m_outputFifo = std::make_unique<FIFO<CSAMPLE>>(
+                m_iNumOutputChannels * framesPerBuffer * 2);
     }
     if (m_iNumInputChannels) {
-        m_inputFifo = std::make_unique<FIFO<CSAMPLE> >(
-                m_iNumInputChannels * m_framesPerBuffer * 2);
+        m_inputFifo = std::make_unique<FIFO<CSAMPLE>>(
+                m_iNumInputChannels * framesPerBuffer * 2);
     }
 
     m_pNetworkStream->startStream(m_dSampleRate);
@@ -89,15 +90,13 @@ SoundDeviceError SoundDeviceNetwork::open(bool isClkRefDevice, int syncBuffers)
         // Update the samplerate and latency ControlObjects, which allow the
         // waveform view to properly correct for the latency.
         ControlObject::set(ConfigKey("[Master]", "latency"),
-                m_audioBufferTime.toDoubleMillis());
+                requestedBufferTime.toDoubleMillis());
         ControlObject::set(ConfigKey("[Master]", "samplerate"), m_dSampleRate);
-        ControlObject::set(ConfigKey("[Master]", "audio_buffer_size"),
-                m_audioBufferTime.toDoubleMillis());
 
         // Network stream was just started above so we have to wait until
         // we can pass one chunk.
         // The first callback runs early to do the one time setups
-        m_targetTime = m_audioBufferTime.toIntegerMicros();
+        m_targetTime = requestedBufferTime.toIntegerMicros();
 
         m_pThread = std::make_unique<SoundDeviceNetworkThread>(this);
         m_pThread->start(QThread::TimeCriticalPriority);
@@ -129,12 +128,12 @@ QString SoundDeviceNetwork::getError() const {
     return QString();
 }
 
-void SoundDeviceNetwork::readProcess() {
+void SoundDeviceNetwork::readProcess(SINT framesPerBuffer) {
     if (!m_inputFifo || !m_pNetworkStream || !m_iNumInputChannels) {
         return;
     }
 
-    int inChunkSize = m_framesPerBuffer * m_iNumInputChannels;
+    int inChunkSize = framesPerBuffer * m_iNumInputChannels;
     int readAvailable = m_pNetworkStream->getReadExpected()
             * m_iNumInputChannels;
     int writeAvailable = m_inputFifo->writeAvailable();
@@ -220,15 +219,15 @@ void SoundDeviceNetwork::readProcess() {
         clearInputBuffer(inChunkSize - readCount, readCount);
     }
 
-    m_pSoundManager->pushInputBuffers(m_audioInputs, m_framesPerBuffer);
+    m_pSoundManager->pushInputBuffers(m_audioInputs, framesPerBuffer);
 }
 
-void SoundDeviceNetwork::writeProcess() {
+void SoundDeviceNetwork::writeProcess(SINT framesPerBuffer) {
     if (!m_outputFifo || !m_pNetworkStream) {
         return;
     }
 
-    int outChunkSize = m_framesPerBuffer * m_iNumOutputChannels;
+    int outChunkSize = framesPerBuffer * m_iNumOutputChannels;
     int writeAvailable = m_outputFifo->writeAvailable();
     int writeCount = outChunkSize;
     if (outChunkSize > writeAvailable) {
@@ -409,8 +408,12 @@ void SoundDeviceNetwork::workerWriteSilence(NetworkOutputStreamWorkerPtr pWorker
 }
 
 void SoundDeviceNetwork::callbackProcessClkRef() {
+    const SINT framesPerBuffer = m_configFramesPerBuffer;
+
     // This must be the very first call, to measure an exact value
-    updateCallbackEntryToDacTime();
+    // NOTE: For network streams the buffer size is always the configured buffer
+    //       size
+    updateCallbackEntryToDacTime(framesPerBuffer);
 
     Trace trace("SoundDeviceNetwork::callbackProcessClkRef %1",
                 m_deviceId.name);
@@ -462,33 +465,34 @@ void SoundDeviceNetwork::callbackProcessClkRef() {
         }
     }
 
-    m_pSoundManager->readProcess();
+    m_pSoundManager->readProcess(framesPerBuffer);
 
     {
         ScopedTimer t("SoundDevicePortAudio::callbackProcess prepare %1",
                 m_deviceId.name);
-        m_pSoundManager->onDeviceOutputCallback(m_framesPerBuffer);
+        m_pSoundManager->onDeviceOutputCallback(framesPerBuffer);
     }
 
-    m_pSoundManager->writeProcess();
+    m_pSoundManager->writeProcess(framesPerBuffer);
 
-    m_pSoundManager->processUnderflowHappened();
+    m_pSoundManager->processUnderflowHappened(framesPerBuffer);
 
-    updateAudioLatencyUsage();
+    updateAudioLatencyUsage(framesPerBuffer);
 }
 
-void SoundDeviceNetwork::updateCallbackEntryToDacTime() {
+void SoundDeviceNetwork::updateCallbackEntryToDacTime(SINT framesPerBuffer) {
     m_clkRefTimer.start();
     qint64 currentTime = m_pNetworkStream->getInputStreamTimeUs();
-    m_targetTime += m_audioBufferTime.toIntegerMicros();
+    // This deadline for the next buffer in microseconds since the Unix epoch
+    m_targetTime += static_cast<qint64>(framesPerBuffer / m_dSampleRate * 1000000);
     double callbackEntrytoDacSecs = (m_targetTime - currentTime) / 1000000.0;
     callbackEntrytoDacSecs = math_max(callbackEntrytoDacSecs, 0.0001);
     VisualPlayPosition::setCallbackEntryToDacSecs(callbackEntrytoDacSecs, m_clkRefTimer);
     //qDebug() << callbackEntrytoDacSecs << timeSinceLastCbSecs;
 }
 
-void SoundDeviceNetwork::updateAudioLatencyUsage() {
-    m_framesSinceAudioLatencyUsageUpdate += m_framesPerBuffer;
+void SoundDeviceNetwork::updateAudioLatencyUsage(SINT framesPerBuffer) {
+    m_framesSinceAudioLatencyUsageUpdate += framesPerBuffer;
     if (m_framesSinceAudioLatencyUsageUpdate
             > (m_dSampleRate / CPU_USAGE_UPDATE_RATE)) {
         double secInAudioCb = m_timeInAudioCallback.toDoubleSeconds();

src/soundio/sounddevicenetwork.h

@@ -32,19 +32,21 @@ class SoundDeviceNetwork : public SoundDevice {
     SoundDeviceError open(bool isClkRefDevice, int syncBuffers) override;
     bool isOpen() const override;
     SoundDeviceError close() override;
-    void readProcess() override;
-    void writeProcess() override;
+    void readProcess(SINT framesPerBuffer) override;
+    void writeProcess(SINT framesPerBuffer) override;
     QString getError() const override;
 
     unsigned int getDefaultSampleRate() const override {
         return 44100;
     }
 
+    // NOTE: This does not take a frames per buffer argument because that is
+    //       always equal to the configured buffer size for network streams
     void callbackProcessClkRef();
 
   private:
-    void updateCallbackEntryToDacTime();
-    void updateAudioLatencyUsage();
+    void updateCallbackEntryToDacTime(SINT framesPerBuffer);
+    void updateAudioLatencyUsage(SINT framesPerBuffer);
 
     void workerWriteProcess(NetworkOutputStreamWorkerPtr pWorker,
             int outChunkSize, int readAvailable,
@@ -61,10 +63,10 @@ class SoundDeviceNetwork : public SoundDevice {
 
     std::unique_ptr<ControlProxy> m_pMasterAudioLatencyUsage;
     mixxx::Duration m_timeInAudioCallback;
-    mixxx::Duration m_audioBufferTime;
     int m_framesSinceAudioLatencyUsageUpdate;
     std::unique_ptr<SoundDeviceNetworkThread> m_pThread;
     bool m_denormals;
+    /// The deadline for the next buffer, in microseconds since the Unix epoch.
     qint64 m_targetTime;
     PerformanceTimer m_clkRefTimer;
 };

src/soundio/sounddevicenotfound.h

@@ -29,9 +29,11 @@ class SoundDeviceNotFound : public SoundDevice {
     SoundDeviceError close() override {
         return SOUNDDEVICE_ERROR_ERR;
     };
-    void readProcess() override { };
-    void writeProcess() override { };
-    QString getError() const override{ return QObject::tr("Device not found"); };
+    void readProcess(SINT /*framesPerbuffer*/) override{};
+    void writeProcess(SINT /*framesPerbuffer*/) override{};
+    QString getError() const override {
+        return QObject::tr("Device not found");
+    };
 
     unsigned int getDefaultSampleRate() const override {
         return 44100;