xref: /dports/audio/audacity/audacity-Audacity-3.1.3/src/AudioIO.cpp

/**********************************************************************

  Audacity: A Digital Audio Editor

  AudioIO.cpp

  Copyright 2000-2004:
  Dominic Mazzoni
  Joshua Haberman
  Markus Meyer
  Matt Brubeck

  This program is free software; you can redistribute it and/or modify it
  under the terms of the GNU General Public License as published by the Free
  Software Foundation; either version 2 of the License, or (at your option)
  any later version.

********************************************************************//**

\class AudioIoCallback
\brief AudioIoCallback is a class that implements the callback required
by PortAudio.  The callback needs to be responsive, has no GUI, and
copies data into and out of the sound card buffers.  It also sends data
to the meters.


*//*****************************************************************//**

\class AudioIO
\brief AudioIO uses the PortAudio library to play and record sound.

  Great care and attention to detail are necessary for understanding and
  modifying this system.  The code in this file is run from three
  different thread contexts: the UI thread, the disk thread (which
  this file creates and maintains; in the code, this is called the
  Audio Thread), and the PortAudio callback thread.
  To highlight this deliniation, the file is divided into three parts
  based on what thread context each function is intended to run in.

  \todo run through all functions called from audio and portaudio threads
  to verify they are thread-safe. Note that synchronization of the style:
  "A sets flag to signal B, B clears flag to acknowledge completion"
  is not thread safe in a general multiple-CPU context. For example,
  B can write to a buffer and set a completion flag. The flag write can
  occur before the buffer write due to out-of-order execution. Then A
  can see the flag and read the buffer before buffer writes complete.

*//****************************************************************//**

\class AudioThread
\brief Defined different on Mac and other platforms (on Mac it does not
use wxWidgets wxThread), this class sits in a thread loop reading and
writing audio.

*//****************************************************************//**

\class AudioIOListener
\brief Monitors record play start/stop and new sample blocks.  Has
callbacks for these events.

*//****************************************************************//**

\class AudioIOStartStreamOptions
\brief struct holding stream options, including a pointer to the
time warp info and AudioIOListener and whether the playback is looped.

*//*******************************************************************/


#include "AudioIO.h"



#include "AudioIOExt.h"
#include "AudioIOListener.h"

#include "float_cast.h"
#include "DeviceManager.h"

#include <cfloat>
#include <math.h>
#include <stdlib.h>
#include <algorithm>
#include <numeric>

#ifdef __WXMSW__
#include <malloc.h>
#endif

#ifdef HAVE_ALLOCA_H
#include <alloca.h>
#endif

#include "portaudio.h"

#if USE_PORTMIXER
#include "portmixer.h"
#endif

#include <wx/app.h>
#include <wx/frame.h>
#include <wx/wxcrtvararg.h>
#include <wx/log.h>
#include <wx/textctrl.h>
#include <wx/timer.h>
#include <wx/intl.h>
#include <wx/debug.h>

#if defined(__WXMAC__) || defined(__WXMSW__)
#include <wx/power.h>
#endif

#include "Meter.h"
#include "Mix.h"
#include "Resample.h"
#include "RingBuffer.h"
#include "Decibels.h"
#include "Prefs.h"
#include "Project.h"
#include "DBConnection.h"
#include "ProjectFileIO.h"
#include "ProjectWindows.h"
#include "WaveTrack.h"

#include "effects/RealtimeEffectManager.h"
#include "QualitySettings.h"
#include "widgets/AudacityMessageBox.h"
#include "BasicUI.h"

#include "Gain.h"

#ifdef EXPERIMENTAL_AUTOMATED_INPUT_LEVEL_ADJUSTMENT
   #define LOWER_BOUND 0.0
   #define UPPER_BOUND 1.0
#endif

using std::max;
using std::min;

AudioIO *AudioIO::Get()
{
   return static_cast< AudioIO* >( AudioIOBase::Get() );
}

wxDEFINE_EVENT(EVT_AUDIOIO_PLAYBACK, wxCommandEvent);
wxDEFINE_EVENT(EVT_AUDIOIO_CAPTURE, wxCommandEvent);
wxDEFINE_EVENT(EVT_AUDIOIO_MONITOR, wxCommandEvent);

// static
int AudioIoCallback::mNextStreamToken = 0;
double AudioIoCallback::mCachedBestRateOut;
bool AudioIoCallback::mCachedBestRatePlaying;
bool AudioIoCallback::mCachedBestRateCapturing;

#ifdef __WXGTK__
   // Might #define this for a useful thing on Linux
   #undef REALTIME_ALSA_THREAD
#else
   // never on the other operating systems
   #undef REALTIME_ALSA_THREAD
#endif

#ifdef REALTIME_ALSA_THREAD
#include "pa_linux_alsa.h"
#endif

int audacityAudioCallback(const void *inputBuffer, void *outputBuffer,
                          unsigned long framesPerBuffer,
                          const PaStreamCallbackTimeInfo *timeInfo,
                          PaStreamCallbackFlags statusFlags, void *userData );


//////////////////////////////////////////////////////////////////////
//
//     class AudioThread - declaration and glue code
//
//////////////////////////////////////////////////////////////////////

#include <thread>

#ifdef __WXMAC__

// On Mac OS X, it's better not to use the wxThread class.
// We use our own implementation based on pthreads instead.

#include <pthread.h>
#include <time.h>

class AudioThread {
 public:
   typedef int ExitCode;
   AudioThread() { mDestroy = false; mThread = NULL; }
   virtual ExitCode Entry();
   void Create() {}
   void Delete() {
      mDestroy = true;
      pthread_join(mThread, NULL);
   }
   bool TestDestroy() { return mDestroy; }
   void Sleep(int ms) {
      struct timespec spec;
      spec.tv_sec = 0;
      spec.tv_nsec = ms * 1000 * 1000;
      nanosleep(&spec, NULL);
   }
   static void *callback(void *p) {
      AudioThread *th = (AudioThread *)p;
      return reinterpret_cast<void *>( th->Entry() );
   }
   void Run() {
      pthread_create(&mThread, NULL, callback, this);
   }
 private:
   bool mDestroy;
   pthread_t mThread;
};

#else

// The normal wxThread-derived AudioThread class for all other
// platforms:
class AudioThread /* not final */ : public wxThread {
 public:
   AudioThread():wxThread(wxTHREAD_JOINABLE) {}
   ExitCode Entry() override;
};

#endif

//////////////////////////////////////////////////////////////////////
//
//     UI Thread Context
//
//////////////////////////////////////////////////////////////////////

void AudioIO::Init()
{
   ugAudioIO.reset(safenew AudioIO());
   Get()->mThread->Run();

   // Make sure device prefs are initialized
   if (gPrefs->Read(wxT("AudioIO/RecordingDevice"), wxT("")).empty()) {
      int i = getRecordDevIndex();
      const PaDeviceInfo *info = Pa_GetDeviceInfo(i);
      if (info) {
         AudioIORecordingDevice.Write(DeviceName(info));
         AudioIOHost.Write(HostName(info));
      }
   }

   if (gPrefs->Read(wxT("AudioIO/PlaybackDevice"), wxT("")).empty()) {
      int i = getPlayDevIndex();
      const PaDeviceInfo *info = Pa_GetDeviceInfo(i);
      if (info) {
         AudioIOPlaybackDevice.Write(DeviceName(info));
         AudioIOHost.Write(HostName(info));
      }
   }

   gPrefs->Flush();
}

void AudioIO::Deinit()
{
   ugAudioIO.reset();
}

bool AudioIO::ValidateDeviceNames(const wxString &play, const wxString &rec)
{
   const PaDeviceInfo *pInfo = Pa_GetDeviceInfo(getPlayDevIndex(play));
   const PaDeviceInfo *rInfo = Pa_GetDeviceInfo(getRecordDevIndex(rec));

   // Valid iff both defined and the same api.
   return pInfo != nullptr && rInfo != nullptr && pInfo->hostApi == rInfo->hostApi;
}

AudioIO::AudioIO()
{
   if (!std::atomic<double>{}.is_lock_free()) {
      // If this check fails, then the atomic<double> members in AudioIO.h
      // might be changed to atomic<float> to be more efficient with some
      // loss of precision.  That could be conditionally compiled depending
      // on the platform.
      wxASSERT(false);
   }

   // This ASSERT because of casting in the callback
   // functions where we cast a tempFloats buffer to a (short*) buffer.
   // We have to ASSERT in the GUI thread, if we are to see it properly.
   wxASSERT( sizeof( short ) <= sizeof( float ));

   mAudioThreadShouldCallTrackBufferExchangeOnce = false;
   mAudioThreadTrackBufferExchangeLoopRunning = false;
   mAudioThreadTrackBufferExchangeLoopActive = false;
   mPortStreamV19 = NULL;

   mNumPauseFrames = 0;

#ifdef EXPERIMENTAL_AUTOMATED_INPUT_LEVEL_ADJUSTMENT
   mAILAActive = false;
#endif
   mStreamToken = 0;

   mLastPaError = paNoError;

   mLastRecordingOffset = 0.0;
   mNumCaptureChannels = 0;
   mPaused = false;
   mSilenceLevel = 0.0;

   mUpdateMeters = false;
   mUpdatingMeters = false;

   mOutputMeter.reset();

   PaError err = Pa_Initialize();

   if (err != paNoError) {
      auto errStr = XO("Could not find any audio devices.\n");
      errStr += XO("You will not be able to play or record audio.\n\n");
      wxString paErrStr = LAT1CTOWX(Pa_GetErrorText(err));
      if (!paErrStr.empty())
         errStr += XO("Error: %s").Format( paErrStr );
      // XXX: we are in libaudacity, popping up dialogs not allowed!  A
      // long-term solution will probably involve exceptions
      AudacityMessageBox(
         errStr,
         XO("Error Initializing Audio"),
         wxICON_ERROR|wxOK);

      // Since PortAudio is not initialized, all calls to PortAudio
      // functions will fail.  This will give reasonable behavior, since
      // the user will be able to do things not relating to audio i/o,
      // but any attempt to play or record will simply fail.
   }

   // Start thread
   mThread = std::make_unique<AudioThread>();
   mThread->Create();

#if defined(USE_PORTMIXER)
   mPortMixer = NULL;
   mPreviousHWPlaythrough = -1.0;
   HandleDeviceChange();
#else
   mEmulateMixerOutputVol = true;
   mInputMixerWorks = false;
#endif

   mMixerOutputVol = AudioIOPlaybackVolume.Read();

   mLastPlaybackTimeMillis = 0;
}

AudioIO::~AudioIO()
{
   if ( !mOwningProject.expired() )
      // Unlikely that this will be destroyed earlier than any projects, but
      // be prepared anyway
      ResetOwningProject();

#if defined(USE_PORTMIXER)
   if (mPortMixer) {
      #if __WXMAC__
      if (Px_SupportsPlaythrough(mPortMixer) && mPreviousHWPlaythrough >= 0.0)
         Px_SetPlaythrough(mPortMixer, mPreviousHWPlaythrough);
         mPreviousHWPlaythrough = -1.0;
      #endif
      Px_CloseMixer(mPortMixer);
      mPortMixer = NULL;
   }
#endif

   // FIXME: ? TRAP_ERR.  Pa_Terminate probably OK if err without reporting.
   Pa_Terminate();

   /* Delete is a "graceful" way to stop the thread.
      (Kill is the not-graceful way.) */

   // This causes reentrancy issues during application shutdown
   // wxTheApp->Yield();

   mThread->Delete();
   mThread.reset();
}

void AudioIO::SetMixer(int inputSource, float recordVolume,
                       float playbackVolume)
{
   mMixerOutputVol = playbackVolume;
   AudioIOPlaybackVolume.Write(mMixerOutputVol);

#if defined(USE_PORTMIXER)
   PxMixer *mixer = mPortMixer;
   if( !mixer )
      return;

   float oldRecordVolume = Px_GetInputVolume(mixer);

   AudioIoCallback::SetMixer(inputSource);
   if( oldRecordVolume != recordVolume )
      Px_SetInputVolume(mixer, recordVolume);

#endif
}

void AudioIO::GetMixer(int *recordDevice, float *recordVolume,
                       float *playbackVolume)
{
   *playbackVolume = mMixerOutputVol;

#if defined(USE_PORTMIXER)

   PxMixer *mixer = mPortMixer;

   if( mixer )
   {
      *recordDevice = Px_GetCurrentInputSource(mixer);

      if (mInputMixerWorks)
         *recordVolume = Px_GetInputVolume(mixer);
      else
         *recordVolume = 1.0f;

      return;
   }

#endif

   *recordDevice = 0;
   *recordVolume = 1.0f;
}

bool AudioIO::InputMixerWorks()
{
   return mInputMixerWorks;
}

wxArrayString AudioIO::GetInputSourceNames()
{
#if defined(USE_PORTMIXER)

   wxArrayString deviceNames;

   if( mPortMixer )
   {
      int numSources = Px_GetNumInputSources(mPortMixer);
      for( int source = 0; source < numSources; source++ )
         deviceNames.push_back(wxString(wxSafeConvertMB2WX(Px_GetInputSourceName(mPortMixer, source))));
   }
   else
   {
      wxLogDebug(wxT("AudioIO::GetInputSourceNames(): PortMixer not initialised!"));
   }

   return deviceNames;

#else

   wxArrayString blank;

   return blank;

#endif
}

static PaSampleFormat AudacityToPortAudioSampleFormat(sampleFormat format)
{
   switch(format) {
   case int16Sample:
      return paInt16;
   case int24Sample:
      return paInt24;
   case floatSample:
   default:
      return paFloat32;
   }
}

bool AudioIO::StartPortAudioStream(const AudioIOStartStreamOptions &options,
                                   unsigned int numPlaybackChannels,
                                   unsigned int numCaptureChannels,
                                   sampleFormat captureFormat)
{
   auto sampleRate = options.rate;
   mNumPauseFrames = 0;
   SetOwningProject( options.pProject );
   bool success = false;
   auto cleanup = finally([&]{
      if (!success)
         ResetOwningProject();
   });

   // PRL:  Protection from crash reported by David Bailes, involving starting
   // and stopping with frequent changes of active window, hard to reproduce
   if (mOwningProject.expired())
      return false;

   mInputMeter.reset();
   mOutputMeter.reset();

   mLastPaError = paNoError;
   // pick a rate to do the audio I/O at, from those available. The project
   // rate is suggested, but we may get something else if it isn't supported
   mRate = GetBestRate(numCaptureChannels > 0, numPlaybackChannels > 0, sampleRate);

   // July 2016 (Carsten and Uwe)
   // BUG 193: Tell PortAudio sound card will handle 24 bit (under DirectSound) using
   // userData.
   int captureFormat_saved = captureFormat;
   // Special case: Our 24-bit sample format is different from PortAudio's
   // 3-byte packed format. So just make PortAudio return float samples,
   // since we need float values anyway to apply the gain.
   // ANSWER-ME: So we *never* actually handle 24-bit?! This causes mCapture to
   // be set to floatSample below.
   // JKC: YES that's right.  Internally Audacity uses float, and float has space for
   // 24 bits as well as exponent.  Actual 24 bit would require packing and
   // unpacking unaligned bytes and would be inefficient.
   // ANSWER ME: is floatSample 64 bit on 64 bit machines?
   if (captureFormat == int24Sample)
      captureFormat = floatSample;

   mNumPlaybackChannels = numPlaybackChannels;
   mNumCaptureChannels = numCaptureChannels;

   bool usePlayback = false, useCapture = false;
   PaStreamParameters playbackParameters{};
   PaStreamParameters captureParameters{};

   auto latencyDuration = AudioIOLatencyDuration.Read();

   if( numPlaybackChannels > 0)
   {
      usePlayback = true;

      // this sets the device index to whatever is "right" based on preferences,
      // then defaults
      playbackParameters.device = getPlayDevIndex();

      const PaDeviceInfo *playbackDeviceInfo;
      playbackDeviceInfo = Pa_GetDeviceInfo( playbackParameters.device );

      if( playbackDeviceInfo == NULL )
         return false;

      // regardless of source formats, we always mix to float
      playbackParameters.sampleFormat = paFloat32;
      playbackParameters.hostApiSpecificStreamInfo = NULL;
      playbackParameters.channelCount = mNumPlaybackChannels;

      if (mSoftwarePlaythrough)
         playbackParameters.suggestedLatency =
            playbackDeviceInfo->defaultLowOutputLatency;
      else {
         // When using WASAPI, the suggested latency does not affect
         // the latency of the playback, but the position of playback is given as if
         // there was the suggested latency. This results in the last "suggested latency"
         // of a selection not being played. So for WASAPI use 0.0 for the suggested
         // latency regardless of user setting. See bug 1949.
         const PaHostApiInfo* hostInfo = Pa_GetHostApiInfo(playbackDeviceInfo->hostApi);
         bool isWASAPI = (hostInfo && hostInfo->type == paWASAPI);
         playbackParameters.suggestedLatency = isWASAPI ? 0.0 : latencyDuration/1000.0;
      }

      mOutputMeter = options.playbackMeter;
   }

   if( numCaptureChannels > 0)
   {
      useCapture = true;
      mCaptureFormat = captureFormat;

      const PaDeviceInfo *captureDeviceInfo;
      // retrieve the index of the device set in the prefs, or a sensible
      // default if it isn't set/valid
      captureParameters.device = getRecordDevIndex();

      captureDeviceInfo = Pa_GetDeviceInfo( captureParameters.device );

      if( captureDeviceInfo == NULL )
         return false;

      captureParameters.sampleFormat =
         AudacityToPortAudioSampleFormat(mCaptureFormat);

      captureParameters.hostApiSpecificStreamInfo = NULL;
      captureParameters.channelCount = mNumCaptureChannels;

      if (mSoftwarePlaythrough)
         captureParameters.suggestedLatency =
            captureDeviceInfo->defaultHighInputLatency;
      else
         captureParameters.suggestedLatency = latencyDuration/1000.0;

      SetCaptureMeter( mOwningProject.lock(), options.captureMeter );
   }

   SetMeters();

#ifdef USE_PORTMIXER
#ifdef __WXMSW__
   //mchinen nov 30 2010.  For some reason Pa_OpenStream resets the input volume on windows.
   //so cache and restore after it.
   //The actual problem is likely in portaudio's pa_win_wmme.c OpenStream().
   float oldRecordVolume = Px_GetInputVolume(mPortMixer);
#endif
#endif

   // July 2016 (Carsten and Uwe)
   // BUG 193: Possibly tell portAudio to use 24 bit with DirectSound.
   int  userData = 24;
   int* lpUserData = (captureFormat_saved == int24Sample) ? &userData : NULL;

   // (Linux, bug 1885) After scanning devices it takes a little time for the
   // ALSA device to be available, so allow retries.
   // On my test machine, no more than 3 attempts are required.
   unsigned int maxTries = 1;
#ifdef __WXGTK__
   if (DeviceManager::Instance()->GetTimeSinceRescan() < 10)
      maxTries = 5;
#endif

   for (unsigned int tries = 0; tries < maxTries; tries++) {
      mLastPaError = Pa_OpenStream( &mPortStreamV19,
                                    useCapture ? &captureParameters : NULL,
                                    usePlayback ? &playbackParameters : NULL,
                                    mRate, paFramesPerBufferUnspecified,
                                    paNoFlag,
                                    audacityAudioCallback, lpUserData );
      if (mLastPaError == paNoError) {
         break;
      }
      wxLogDebug("Attempt %u to open capture stream failed with: %d", 1 + tries, mLastPaError);
      wxMilliSleep(1000);
   }


#if USE_PORTMIXER
#ifdef __WXMSW__
   Px_SetInputVolume(mPortMixer, oldRecordVolume);
#endif
   if (mPortStreamV19 != NULL && mLastPaError == paNoError) {

      #ifdef __WXMAC__
      if (mPortMixer) {
         if (Px_SupportsPlaythrough(mPortMixer)) {
            bool playthrough = false;

            mPreviousHWPlaythrough = Px_GetPlaythrough(mPortMixer);

            // Bug 388.  Feature not supported.
            //gPrefs->Read(wxT("/AudioIO/Playthrough"), &playthrough, false);
            if (playthrough)
               Px_SetPlaythrough(mPortMixer, 1.0);
            else
               Px_SetPlaythrough(mPortMixer, 0.0);
         }
      }
      #endif
   }
#endif

#if (defined(__WXMAC__) || defined(__WXMSW__)) && wxCHECK_VERSION(3,1,0)
   // Don't want the system to sleep while audio I/O is active
   if (mPortStreamV19 != NULL && mLastPaError == paNoError) {
      wxPowerResource::Acquire(wxPOWER_RESOURCE_SCREEN, _("Audacity Audio"));
   }
#endif

   return (success = (mLastPaError == paNoError));
}

wxString AudioIO::LastPaErrorString()
{
   return wxString::Format(wxT("%d %s."), (int) mLastPaError, Pa_GetErrorText(mLastPaError));
}

void AudioIO::SetOwningProject(
   const std::shared_ptr<AudacityProject> &pProject )
{
   if ( !mOwningProject.expired() ) {
      wxASSERT(false);
      ResetOwningProject();
   }

   mOwningProject = pProject;
}

void AudioIO::ResetOwningProject()
{
   mOwningProject.reset();
}

void AudioIO::StartMonitoring( const AudioIOStartStreamOptions &options )
{
   if ( mPortStreamV19 || mStreamToken )
      return;

   bool success;
   auto captureFormat = QualitySettings::SampleFormatChoice();
   auto captureChannels = AudioIORecordChannels.Read();
   gPrefs->Read(wxT("/AudioIO/SWPlaythrough"), &mSoftwarePlaythrough, false);
   int playbackChannels = 0;

   if (mSoftwarePlaythrough)
      playbackChannels = 2;

   // FIXME: TRAP_ERR StartPortAudioStream (a PaError may be present)
   // but StartPortAudioStream function only returns true or false.
   mUsingAlsa = false;
   success = StartPortAudioStream(options, (unsigned int)playbackChannels,
                                  (unsigned int)captureChannels,
                                  captureFormat);

   auto pOwningProject = mOwningProject.lock();
   if (!success) {
      using namespace BasicUI;
      auto msg = XO("Error opening recording device.\nError code: %s")
         .Format( Get()->LastPaErrorString() );
      ShowErrorDialog( *ProjectFramePlacement( pOwningProject.get() ),
         XO("Error"), msg, wxT("Error_opening_sound_device"),
         ErrorDialogOptions{ ErrorDialogType::ModalErrorReport } );
      return;
   }

   wxCommandEvent e(EVT_AUDIOIO_MONITOR);
   e.SetEventObject( pOwningProject.get() );
   e.SetInt(true);
   wxTheApp->ProcessEvent(e);

   // FIXME: TRAP_ERR PaErrorCode 'noted' but not reported in StartMonitoring.
   // Now start the PortAudio stream!
   // TODO: ? Factor out and reuse error reporting code from end of
   // AudioIO::StartStream?
   mLastPaError = Pa_StartStream( mPortStreamV19 );

   // Update UI display only now, after all possibilities for error are past.
   auto pListener = GetListener();
   if ((mLastPaError == paNoError) && pListener) {
      // advertise the chosen I/O sample rate to the UI
      pListener->OnAudioIORate((int)mRate);
   }
}

int AudioIO::StartStream(const TransportTracks &tracks,
   double t0, double t1, double mixerLimit,
   const AudioIOStartStreamOptions &options)
{
   const auto &pStartTime = options.pStartTime;
   t1 = std::min(t1, mixerLimit);

   mLostSamples = 0;
   mLostCaptureIntervals.clear();
   mDetectDropouts =
      gPrefs->Read( WarningDialogKey(wxT("DropoutDetected")), true ) != 0;
   auto cleanup = finally ( [this] { ClearRecordingException(); } );

   if( IsBusy() )
      return 0;

   // We just want to set mStreamToken to -1 - this way avoids
   // an extremely rare but possible race condition, if two functions
   // somehow called StartStream at the same time...
   mStreamToken--;
   if (mStreamToken != -1)
      return 0;

   // TODO: we don't really need to close and reopen stream if the
   // format matches; however it's kind of tricky to keep it open...
   //
   //   if (sampleRate == mRate &&
   //       playbackChannels == mNumPlaybackChannels &&
   //       captureChannels == mNumCaptureChannels &&
   //       captureFormat == mCaptureFormat) {

   if (mPortStreamV19) {
      StopStream();
      while(mPortStreamV19)
         wxMilliSleep( 50 );
   }

#ifdef __WXGTK__
   // Detect whether ALSA is the chosen host, and do the various involved MIDI
   // timing compensations only then.
   mUsingAlsa = (AudioIOHost.Read() == L"ALSA");
#endif

   gPrefs->Read(wxT("/AudioIO/SWPlaythrough"), &mSoftwarePlaythrough, false);
   gPrefs->Read(wxT("/AudioIO/SoundActivatedRecord"), &mPauseRec, false);
   gPrefs->Read(wxT("/AudioIO/Microfades"), &mbMicroFades, false);
   int silenceLevelDB;
   gPrefs->Read(wxT("/AudioIO/SilenceLevel"), &silenceLevelDB, -50);
   int dBRange = DecibelScaleCutoff.Read();
   if(silenceLevelDB < -dBRange)
   {
      silenceLevelDB = -dBRange + 3;
      // meter range was made smaller than SilenceLevel
      // so set SilenceLevel reasonable

      // PRL:  update prefs, or correct it only in-session?
      // The behavior (as of 2.3.1) was the latter, the code suggested that
      // the intent was the former;  I preserve the behavior, but uncomment
      // this if you disagree.
      // gPrefs->Write(wxT("/AudioIO/SilenceLevel"), silenceLevelDB);
      // gPrefs->Flush();
   }
   mSilenceLevel = DB_TO_LINEAR(silenceLevelDB);  // meter goes -dBRange dB -> 0dB

   // Clamp pre-roll so we don't play before time 0
   const auto preRoll = std::max(0.0, std::min(t0, options.preRoll));
   mRecordingSchedule = {};
   mRecordingSchedule.mPreRoll = preRoll;
   mRecordingSchedule.mLatencyCorrection =
      AudioIOLatencyCorrection.Read() / 1000.0;
   mRecordingSchedule.mDuration = t1 - t0;
   if (options.pCrossfadeData)
      mRecordingSchedule.mCrossfadeData.swap( *options.pCrossfadeData );

   mListener = options.listener;
   mRate    = options.rate;

   mSeek    = 0;
   mLastRecordingOffset = 0;
   mCaptureTracks = tracks.captureTracks;
   mPlaybackTracks = tracks.playbackTracks;

   bool commit = false;
   auto cleanupTracks = finally([&]{
      if (!commit) {
         // Don't keep unnecessary shared pointers to tracks
         mPlaybackTracks.clear();
         mCaptureTracks.clear();
         for(auto &ext : Extensions())
            ext.AbortOtherStream();

         // Don't cause a busy wait in the audio thread after stopping scrubbing
         mPlaybackSchedule.ResetMode();
      }
   });

   mPlaybackBuffers.reset();
   mPlaybackMixers.clear();
   mCaptureBuffers.reset();
   mResample.reset();
   mPlaybackSchedule.mTimeQueue.Clear();

   mPlaybackSchedule.Init(
      t0, t1, options, mCaptureTracks.empty() ? nullptr : &mRecordingSchedule );

   unsigned int playbackChannels = 0;
   unsigned int captureChannels = 0;
   sampleFormat captureFormat = floatSample;

   auto pListener = GetListener();

   if (tracks.playbackTracks.size() > 0
      || tracks.otherPlayableTracks.size() > 0)
      playbackChannels = 2;

   if (mSoftwarePlaythrough)
      playbackChannels = 2;

   if (tracks.captureTracks.size() > 0)
   {
      // For capture, every input channel gets its own track
      captureChannels = mCaptureTracks.size();
      // I don't deal with the possibility of the capture tracks
      // having different sample formats, since it will never happen
      // with the current code.  This code wouldn't *break* if this
      // assumption was false, but it would be sub-optimal.  For example,
      // if the first track was 16-bit and the second track was 24-bit,
      // we would set the sound card to capture in 16 bits and the second
      // track wouldn't get the benefit of all 24 bits the card is capable
      // of.
      captureFormat = mCaptureTracks[0]->GetSampleFormat();

      // Tell project that we are about to start recording
      if (pListener)
         pListener->OnAudioIOStartRecording();
   }

   bool successAudio;

   successAudio = StartPortAudioStream(options, playbackChannels,
                                       captureChannels, captureFormat);

   // Call this only after reassignment of mRate that might happen in the
   // previous call.
   mPlaybackSchedule.GetPolicy().Initialize( mPlaybackSchedule, mRate );

#ifdef EXPERIMENTAL_MIDI_OUT
   auto range = Extensions();
   successAudio = successAudio &&
      std::all_of(range.begin(), range.end(),
         [this, &tracks, t0](auto &ext){
            return ext.StartOtherStream( tracks,
              (mPortStreamV19 != NULL && mLastPaError == paNoError)
                 ? Pa_GetStreamInfo(mPortStreamV19) : nullptr,
              t0, mRate ); });
#endif

   if (!successAudio) {
      if (pListener && captureChannels > 0)
         pListener->OnAudioIOStopRecording();
      mStreamToken = 0;

      return 0;
   }

   {
      double mixerStart = t0;
      if (pStartTime)
         mixerStart = std::min( mixerStart, *pStartTime );
      if ( ! AllocateBuffers( options, tracks,
         mixerStart, mixerLimit, options.rate ) )
         return 0;
   }

   if (mNumPlaybackChannels > 0)
   {
      auto & em = RealtimeEffectManager::Get();
      // Setup for realtime playback at the rate of the realtime
      // stream, not the rate of the track.
      em.RealtimeInitialize(mRate);

      // The following adds a NEW effect processor for each logical track and the
      // group determination should mimic what is done in audacityAudioCallback()
      // when calling RealtimeProcess().
      int group = 0;
      for (size_t i = 0, cnt = mPlaybackTracks.size(); i < cnt;)
      {
         const WaveTrack *vt = mPlaybackTracks[i].get();

         // TODO: more-than-two-channels
         unsigned chanCnt = TrackList::Channels(vt).size();
         i += chanCnt;

         // Setup for realtime playback at the rate of the realtime
         // stream, not the rate of the track.
         em.RealtimeAddProcessor(group++, std::min(2u, chanCnt), mRate);
      }
   }

#ifdef EXPERIMENTAL_AUTOMATED_INPUT_LEVEL_ADJUSTMENT
   AILASetStartTime();
#endif

   if (pStartTime)
   {
      // Calculate the NEW time position
      const auto time = *pStartTime;

      // Main thread's initialization of mTime
      mPlaybackSchedule.SetTrackTime( time );
      mPlaybackSchedule.GetPolicy().OffsetTrackTime( mPlaybackSchedule, 0 );

      // Reset mixer positions for all playback tracks
      unsigned numMixers = mPlaybackTracks.size();
      for (unsigned ii = 0; ii < numMixers; ++ii)
         mPlaybackMixers[ii]->Reposition( time );
   }

   // Now that we are done with AllocateBuffers() and SetTrackTime():
   mPlaybackSchedule.mTimeQueue.Prime(mPlaybackSchedule.GetTrackTime());
   // else recording only without overdub

   // We signal the audio thread to call TrackBufferExchange, to prime the RingBuffers
   // so that they will have data in them when the stream starts.  Having the
   // audio thread call TrackBufferExchange here makes the code more predictable, since
   // TrackBufferExchange will ALWAYS get called from the Audio thread.
   mAudioThreadShouldCallTrackBufferExchangeOnce = true;

   while( mAudioThreadShouldCallTrackBufferExchangeOnce ) {
      auto interval = 50ull;
      if (options.playbackStreamPrimer) {
         interval = options.playbackStreamPrimer();
      }
      wxMilliSleep( interval );
   }

   if(mNumPlaybackChannels > 0 || mNumCaptureChannels > 0) {

#ifdef REALTIME_ALSA_THREAD
      // PRL: Do this in hope of less thread scheduling jitter in calls to
      // audacityAudioCallback.
      // Not needed to make audio playback work smoothly.
      // But needed in case we also play MIDI, so that the variable "offset"
      // in AudioIO::MidiTime() is a better approximation of the duration
      // between the call of audacityAudioCallback and the actual output of
      // the first audio sample.
      // (Which we should be able to determine from fields of
      // PaStreamCallbackTimeInfo, but that seems not to work as documented with
      // ALSA.)
      if (mUsingAlsa)
         // Perhaps we should do this only if also playing MIDI ?
         PaAlsa_EnableRealtimeScheduling( mPortStreamV19, 1 );
#endif

      //
      // Generate a unique value each time, to be returned to
      // clients accessing the AudioIO API, so they can query if they
      // are the ones who have reserved AudioIO or not.
      //
      // It is important to set this before setting the portaudio stream in
      // motion -- otherwise it may play an unspecified number of leading
      // zeroes.
      mStreamToken = (++mNextStreamToken);

      // This affects the AudioThread (not the portaudio callback).
      // Probably not needed so urgently before portaudio thread start for usual
      // playback, since our ring buffers have been primed already with 4 sec
      // of audio, but then we might be scrubbing, so do it.
      mAudioThreadTrackBufferExchangeLoopRunning = true;
      mForceFadeOut.store(false, std::memory_order_relaxed);

      // Now start the PortAudio stream!
      PaError err;
      err = Pa_StartStream( mPortStreamV19 );

      if( err != paNoError )
      {
         mStreamToken = 0;
         mAudioThreadTrackBufferExchangeLoopRunning = false;
         if (pListener && mNumCaptureChannels > 0)
            pListener->OnAudioIOStopRecording();
         StartStreamCleanup();
         // PRL: PortAudio error messages are sadly not internationalized
         AudacityMessageBox(
            Verbatim( LAT1CTOWX(Pa_GetErrorText(err)) ) );
         return 0;
      }
   }

   // Update UI display only now, after all possibilities for error are past.
   if (pListener) {
      // advertise the chosen I/O sample rate to the UI
      pListener->OnAudioIORate((int)mRate);
   }

   auto pOwningProject = mOwningProject.lock();
   if (mNumPlaybackChannels > 0)
   {
      wxCommandEvent e(EVT_AUDIOIO_PLAYBACK);
      e.SetEventObject( pOwningProject.get() );
      e.SetInt(true);
      wxTheApp->ProcessEvent(e);
   }

   if (mNumCaptureChannels > 0)
   {
      wxCommandEvent e(EVT_AUDIOIO_CAPTURE);
      e.SetEventObject( pOwningProject.get() );
      e.SetInt(true);
      wxTheApp->ProcessEvent(e);
   }

   commit = true;
   return mStreamToken;
}

void AudioIO::DelayActions(bool recording)
{
   mDelayingActions = recording;
}

bool AudioIO::DelayingActions() const
{
   return mDelayingActions || (mPortStreamV19 && mNumCaptureChannels > 0);
}

void AudioIO::CallAfterRecording(PostRecordingAction action)
{
   if (!action)
      return;

   {
      std::lock_guard<std::mutex> guard{ mPostRecordingActionMutex };
      if (mPostRecordingAction) {
         // Enqueue it, even if perhaps not still recording,
         // but it wasn't cleared yet
         mPostRecordingAction = [
            prevAction = std::move(mPostRecordingAction),
            nextAction = std::move(action)
         ]{ prevAction(); nextAction(); };
         return;
      }
      else if (DelayingActions()) {
         mPostRecordingAction = std::move(action);
         return;
      }
   }

   // Don't delay it except until idle time.
   // (Recording might start between now and then, but won't go far before
   // the action is done.  So the system isn't bulletproof yet.)
   wxTheApp->CallAfter(std::move(action));
}

bool AudioIO::AllocateBuffers(
   const AudioIOStartStreamOptions &options,
   const TransportTracks &tracks, double t0, double t1, double sampleRate )
{
   bool success = false;
   auto cleanup = finally([&]{
      if (!success) StartStreamCleanup( false );
   });

   auto &policy = mPlaybackSchedule.GetPolicy();
   auto times = policy.SuggestedBufferTimes(mPlaybackSchedule);

   //
   // The (audio) stream has been opened successfully (assuming we tried
   // to open it). We now proceed to
   // allocate the memory structures the stream will need.
   //

   //
   // The RingBuffer sizes, and the max amount of the buffer to
   // fill at a time, both grow linearly with the number of
   // tracks.  This allows us to scale up to many tracks without
   // killing performance.
   //

   // real playback time to produce with each filling of the buffers
   // by the Audio thread (except at the end of playback):
   // usually, make fillings fewer and longer for less CPU usage.
   // What Audio thread produces for playback is then consumed by the PortAudio
   // thread, in many smaller pieces.
   double playbackTime = lrint(times.batchSize * mRate) / mRate;

   wxASSERT( playbackTime >= 0 );
   mPlaybackSamplesToCopy = playbackTime * mRate;

   // Capacity of the playback buffer.
   mPlaybackRingBufferSecs = times.ringBufferDelay;

   mCaptureRingBufferSecs =
      4.5 + 0.5 * std::min(size_t(16), mCaptureTracks.size());
   mMinCaptureSecsToCopy =
      0.2 + 0.2 * std::min(size_t(16), mCaptureTracks.size());

   bool bDone;
   do
   {
      bDone = true; // assume success
      try
      {
         if( mNumPlaybackChannels > 0 ) {
            // Allocate output buffers.  For every output track we allocate
            // a ring buffer of ten seconds
            auto playbackBufferSize =
               (size_t)lrint(mRate * mPlaybackRingBufferSecs);

            // Always make at least one playback buffer
            mPlaybackBuffers.reinit(
               std::max<size_t>(1, mPlaybackTracks.size()));
            mPlaybackMixers.clear();
            mPlaybackMixers.resize(mPlaybackTracks.size());

            const auto &warpOptions =
               policy.MixerWarpOptions(mPlaybackSchedule);

            mPlaybackQueueMinimum = lrint( mRate * times.latency );
            mPlaybackQueueMinimum =
               std::min( mPlaybackQueueMinimum, playbackBufferSize );

            if (mPlaybackTracks.empty())
               // Make at least one playback buffer
               mPlaybackBuffers[0] =
                  std::make_unique<RingBuffer>(floatSample, playbackBufferSize);

            for (unsigned int i = 0; i < mPlaybackTracks.size(); i++)
            {
               // Bug 1763 - We must fade in from zero to avoid a click on starting.
               mPlaybackTracks[i]->SetOldChannelGain(0, 0.0);
               mPlaybackTracks[i]->SetOldChannelGain(1, 0.0);

               mPlaybackBuffers[i] =
                  std::make_unique<RingBuffer>(floatSample, playbackBufferSize);

               // use track time for the end time, not real time!
               WaveTrackConstArray mixTracks;
               mixTracks.push_back(mPlaybackTracks[i]);

               double startTime, endTime;
               if (make_iterator_range(tracks.prerollTracks)
                      .contains(mPlaybackTracks[i])) {
                  // Stop playing this track after pre-roll
                  startTime = mPlaybackSchedule.mT0;
                  endTime = t0;
               }
               else {
                  // Pass t1 -- not mT1 as may have been adjusted for latency
                  // -- so that overdub recording stops playing back samples
                  // at the right time, though transport may continue to record
                  startTime = t0;
                  endTime = t1;
               }

               mPlaybackMixers[i] = std::make_unique<Mixer>
                  (mixTracks,
                  // Don't throw for read errors, just play silence:
                  false,
                  warpOptions,
                  startTime,
                  endTime,
                  1,
                  std::max( mPlaybackSamplesToCopy, mPlaybackQueueMinimum ),
                  false,
                  mRate, floatSample,
                  false, // low quality dithering and resampling
                  nullptr,
                  false // don't apply track gains
               );
            }

            const auto timeQueueSize = 1 +
               (playbackBufferSize + TimeQueueGrainSize - 1)
                  / TimeQueueGrainSize;
            mPlaybackSchedule.mTimeQueue.Resize( timeQueueSize );
         }

         if( mNumCaptureChannels > 0 )
         {
            // Allocate input buffers.  For every input track we allocate
            // a ring buffer of five seconds
            auto captureBufferSize =
               (size_t)(mRate * mCaptureRingBufferSecs + 0.5);

            // In the extraordinarily rare case that we can't even afford
            // 100 samples, just give up.
            if(captureBufferSize < 100)
            {
               AudacityMessageBox( XO("Out of memory!") );
               return false;
            }

            mCaptureBuffers.reinit(mCaptureTracks.size());
            mResample.reinit(mCaptureTracks.size());
            mFactor = sampleRate / mRate;

            for( unsigned int i = 0; i < mCaptureTracks.size(); i++ )
            {
               mCaptureBuffers[i] = std::make_unique<RingBuffer>(
                  mCaptureTracks[i]->GetSampleFormat(), captureBufferSize );
               mResample[i] =
                  std::make_unique<Resample>(true, mFactor, mFactor);
                  // constant rate resampling
            }
         }
      }
      catch(std::bad_alloc&)
      {
         // Oops!  Ran out of memory.  This is pretty rare, so we'll just
         // try deleting everything, halving our buffer size, and try again.
         StartStreamCleanup(true);
         mPlaybackRingBufferSecs *= 0.5;
         mPlaybackSamplesToCopy /= 2;
         mCaptureRingBufferSecs *= 0.5;
         mMinCaptureSecsToCopy *= 0.5;
         bDone = false;

         // In the extraordinarily rare case that we can't even afford 100
         // samples, just give up.
         auto playbackBufferSize =
            (size_t)lrint(mRate * mPlaybackRingBufferSecs);
         if(playbackBufferSize < 100 || mPlaybackSamplesToCopy < 100)
         {
            AudacityMessageBox( XO("Out of memory!") );
            return false;
         }
      }
   } while(!bDone);

   success = true;
   return true;
}

void AudioIO::StartStreamCleanup(bool bOnlyBuffers)
{
   if (mNumPlaybackChannels > 0)
   {
      RealtimeEffectManager::Get().RealtimeFinalize();
   }

   mPlaybackBuffers.reset();
   mPlaybackMixers.clear();
   mCaptureBuffers.reset();
   mResample.reset();
   mPlaybackSchedule.mTimeQueue.Clear();

   if(!bOnlyBuffers)
   {
      Pa_AbortStream( mPortStreamV19 );
      Pa_CloseStream( mPortStreamV19 );
      mPortStreamV19 = NULL;
      mStreamToken = 0;
   }

   mPlaybackSchedule.GetPolicy().Finalize( mPlaybackSchedule );
}

bool AudioIO::IsAvailable(AudacityProject &project) const
{
   auto pOwningProject = mOwningProject.lock();
   return !pOwningProject || pOwningProject.get() == &project;
}

void AudioIO::SetMeters()
{
   if (auto pInputMeter = mInputMeter.lock())
      pInputMeter->Reset(mRate, true);
   if (auto pOutputMeter = mOutputMeter.lock())
      pOutputMeter->Reset(mRate, true);

   mUpdateMeters = true;
}

void AudioIO::StopStream()
{
   auto cleanup = finally ( [this] {
      ClearRecordingException();
      mRecordingSchedule.mCrossfadeData.clear(); // free arrays
   } );

   if( mPortStreamV19 == NULL )
      return;

   // DV: This code seems to be unnecessary.
   // We do not leave mPortStreamV19 open in stopped
   // state. (Do we?)
   // This breaks WASAPI backend, as it sets the `running`
   // flag to `false` asynchronously.
   // Previously we have patched PortAudio and the patch
   // was breaking IsStreamStopped() == !IsStreamActive()
   // invariant.
   /*
   if ( Pa_IsStreamStopped(mPortStreamV19) )
      return;
   */

#if (defined(__WXMAC__) || defined(__WXMSW__)) && wxCHECK_VERSION(3,1,0)
   // Re-enable system sleep
   wxPowerResource::Release(wxPOWER_RESOURCE_SCREEN);
#endif

   if( mAudioThreadTrackBufferExchangeLoopRunning )
   {
      // PortAudio callback can use the information that we are stopping to fade
      // out the audio.  Give PortAudio callback a chance to do so.
      mForceFadeOut.store(true, std::memory_order_relaxed);
      auto latency = static_cast<long>(AudioIOLatencyDuration.Read());
      // If we can gracefully fade out in 200ms, with the faded-out play buffers making it through
      // the sound card, then do so.  If we can't, don't wait around.  Just stop quickly and accept
      // there will be a click.
      if( mbMicroFades  && (latency < 150 ))
         wxMilliSleep( latency + 50);
   }

   wxMutexLocker locker(mSuspendAudioThread);

   // No longer need effects processing
   if (mNumPlaybackChannels > 0)
   {
      RealtimeEffectManager::Get().RealtimeFinalize();
   }

   //
   // We got here in one of two ways:
   //
   // 1. The user clicked the stop button and we therefore want to stop
   //    as quickly as possible.  So we use AbortStream().  If this is
   //    the case the portaudio stream is still in the Running state
   //    (see PortAudio state machine docs).
   //
   // 2. The callback told PortAudio to stop the stream since it had
   //    reached the end of the selection.  The UI thread discovered
   //    this by noticing that AudioIO::IsActive() returned false.
   //    IsActive() (which calls Pa_GetStreamActive()) will not return
   //    false until all buffers have finished playing, so we can call
   //    AbortStream without losing any samples.  If this is the case
   //    we are in the "callback finished state" (see PortAudio state
   //    machine docs).
   //
   // The moral of the story: We can call AbortStream safely, without
   // losing samples.
   //
   // DMM: This doesn't seem to be true; it seems to be necessary to
   // call StopStream if the callback brought us here, and AbortStream
   // if the user brought us here.
   //
   // DV: Seems that Pa_CloseStream calls Pa_AbortStream internally,
   // at least for PortAudio 19.7.0+

   mAudioThreadTrackBufferExchangeLoopRunning = false;

   // Audacity can deadlock if it tries to update meters while
   // we're stopping PortAudio (because the meter updating code
   // tries to grab a UI mutex while PortAudio tries to join a
   // pthread).  So we tell the callback to stop updating meters,
   // and wait until the callback has left this part of the code
   // if it was already there.
   mUpdateMeters = false;
   while(mUpdatingMeters) {
      ::wxSafeYield();
      wxMilliSleep( 50 );
   }

   // Turn off HW playthrough if PortMixer is being used

  #if defined(USE_PORTMIXER)
   if( mPortMixer ) {
      #if __WXMAC__
      if (Px_SupportsPlaythrough(mPortMixer) && mPreviousHWPlaythrough >= 0.0)
         Px_SetPlaythrough(mPortMixer, mPreviousHWPlaythrough);
         mPreviousHWPlaythrough = -1.0;
      #endif
   }
  #endif

   if (mPortStreamV19) {
      // DV: Pa_CloseStream will close Pa_AbortStream internally,
      // but it doesn't hurt to do it ourselves.
      // PA_AbortStream will silently fail if stream is stopped.
      if (!Pa_IsStreamStopped( mPortStreamV19 ))
        Pa_AbortStream( mPortStreamV19 );

      Pa_CloseStream( mPortStreamV19 );

      mPortStreamV19 = NULL;
   }

   for( auto &ext : Extensions() )
      ext.StopOtherStream();

   auto pListener = GetListener();

   // If there's no token, we were just monitoring, so we can
   // skip this next part...
   if (mStreamToken > 0) {
      // In either of the above cases, we want to make sure that any
      // capture data that made it into the PortAudio callback makes it
      // to the target WaveTrack.  To do this, we ask the audio thread to
      // call TrackBufferExchange one last time (it normally would not do so since
      // Pa_GetStreamActive() would now return false
      mAudioThreadShouldCallTrackBufferExchangeOnce = true;

      while( mAudioThreadShouldCallTrackBufferExchangeOnce )
      {
         //FIXME: Seems like this block of the UI thread isn't bounded,
         //but we cannot allow event handlers to see incompletely terminated
         //AudioIO state with wxYield (or similar functions)
         wxMilliSleep( 50 );
      }

      //
      // Everything is taken care of.  Now, just free all the resources
      // we allocated in StartStream()
      //

      if (mPlaybackTracks.size() > 0)
      {
         mPlaybackBuffers.reset();
         mPlaybackMixers.clear();
         mPlaybackSchedule.mTimeQueue.Clear();
      }

      //
      // Offset all recorded tracks to account for latency
      //
      if (mCaptureTracks.size() > 0)
      {
         mCaptureBuffers.reset();
         mResample.reset();

         //
         // We only apply latency correction when we actually played back
         // tracks during the recording. If we did not play back tracks,
         // there's nothing we could be out of sync with. This also covers the
         // case that we do not apply latency correction when recording the
         // first track in a project.
         //

         for (unsigned int i = 0; i < mCaptureTracks.size(); i++) {
            // The calls to Flush
            // may cause exceptions because of exhaustion of disk space.
            // Stop those exceptions here, or else they propagate through too
            // many parts of Audacity that are not effects or editing
            // operations.  GuardedCall ensures that the user sees a warning.

            // Also be sure to Flush each track, at the top of the guarded call,
            // relying on the guarantee that the track will be left in a flushed
            // state, though the append buffer may be lost.

            GuardedCall( [&] {
               WaveTrack* track = mCaptureTracks[i].get();

               // use No-fail-guarantee that track is flushed,
               // Partial-guarantee that some initial length of the recording
               // is saved.
               // See comments in TrackBufferExchange().
               track->Flush();
            } );
         }


         if (!mLostCaptureIntervals.empty())
         {
            // This scope may combine many splittings of wave tracks
            // into one transaction, lessening the number of checkpoints
            Optional<TransactionScope> pScope;
            auto pOwningProject = mOwningProject.lock();
            if (pOwningProject) {
               auto &pIO = ProjectFileIO::Get(*pOwningProject);
               pScope.emplace(pIO.GetConnection(), "Dropouts");
            }
            for (auto &interval : mLostCaptureIntervals) {
               auto &start = interval.first;
               auto duration = interval.second;
               for (auto &track : mCaptureTracks) {
                  GuardedCall([&] {
                     track->SyncLockAdjust(start, start + duration);
                  });
               }
            }
            if (pScope)
               pScope->Commit();
         }

         if (pListener)
            pListener->OnCommitRecording();
      }
   }

   if (auto pInputMeter = mInputMeter.lock())
      pInputMeter->Reset(mRate, false);

   if (auto pOutputMeter = mOutputMeter.lock())
      pOutputMeter->Reset(mRate, false);

   mInputMeter.reset();
   mOutputMeter.reset();
   ResetOwningProject();

   if (pListener && mNumCaptureChannels > 0)
      pListener->OnAudioIOStopRecording();

   wxTheApp->CallAfter([this]{
      if (mPortStreamV19 && mNumCaptureChannels > 0)
         // Recording was restarted between StopStream and idle time
         // So the actions can keep waiting
         return;
      // In case some other thread was waiting on the mutex too:
      std::this_thread::yield();
      std::lock_guard<std::mutex> guard{ mPostRecordingActionMutex };
      if (mPostRecordingAction) {
         mPostRecordingAction();
         mPostRecordingAction = {};
      }
      DelayActions(false);
   });

   //
   // Only set token to 0 after we're totally finished with everything
   //
   bool wasMonitoring = mStreamToken == 0;
   mStreamToken = 0;

   if (mNumPlaybackChannels > 0)
   {
      wxCommandEvent e(EVT_AUDIOIO_PLAYBACK);
      auto pOwningProject = mOwningProject.lock();
      e.SetEventObject(pOwningProject.get());
      e.SetInt(false);
      wxTheApp->ProcessEvent(e);
   }

   if (mNumCaptureChannels > 0)
   {
      wxCommandEvent e(wasMonitoring ? EVT_AUDIOIO_MONITOR : EVT_AUDIOIO_CAPTURE);
      auto pOwningProject = mOwningProject.lock();
      e.SetEventObject(pOwningProject.get());
      e.SetInt(false);
      wxTheApp->ProcessEvent(e);
   }

   mNumCaptureChannels = 0;
   mNumPlaybackChannels = 0;

   mPlaybackTracks.clear();
   mCaptureTracks.clear();

   mPlaybackSchedule.GetPolicy().Finalize( mPlaybackSchedule );

   if (pListener) {
      // Tell UI to hide sample rate
      pListener->OnAudioIORate(0);
   }

   // Don't cause a busy wait in the audio thread after stopping scrubbing
   mPlaybackSchedule.ResetMode();
}

void AudioIO::SetPaused(bool state)
{
   if (state != mPaused)
   {
      if (state)
      {
         RealtimeEffectManager::Get().RealtimeSuspend();
      }
      else
      {
         RealtimeEffectManager::Get().RealtimeResume();
      }
   }

   mPaused = state;
}

double AudioIO::GetBestRate(bool capturing, bool playing, double sampleRate)
{
   // Check if we can use the cached value
   if (mCachedBestRateIn != 0.0 && mCachedBestRateIn == sampleRate
      && mCachedBestRatePlaying == playing && mCachedBestRateCapturing == capturing) {
      return mCachedBestRateOut;
   }

   // In order to cache the value, all early returns should instead set retval
   // and jump to finished
   double retval;

   std::vector<long> rates;
   if (capturing) wxLogDebug(wxT("AudioIO::GetBestRate() for capture"));
   if (playing) wxLogDebug(wxT("AudioIO::GetBestRate() for playback"));
   wxLogDebug(wxT("GetBestRate() suggested rate %.0lf Hz"), sampleRate);

   if (capturing && !playing) {
      rates = GetSupportedCaptureRates(-1, sampleRate);
   }
   else if (playing && !capturing) {
      rates = GetSupportedPlaybackRates(-1, sampleRate);
   }
   else {   // we assume capturing and playing - the alternative would be a
            // bit odd
      rates = GetSupportedSampleRates(-1, -1, sampleRate);
   }
   /* rem rates is the array of hardware-supported sample rates (in the current
    * configuration), sampleRate is the Project Rate (desired sample rate) */
   long rate = (long)sampleRate;

   if (make_iterator_range(rates).contains(rate)) {
      wxLogDebug(wxT("GetBestRate() Returning %.0ld Hz"), rate);
      retval = rate;
      goto finished;
      /* the easy case - the suggested rate (project rate) is in the list, and
       * we can just accept that and send back to the caller. This should be
       * the case for most users most of the time (all of the time on
       * Win MME as the OS does resampling) */
   }

   /* if we get here, there is a problem - the project rate isn't supported
    * on our hardware, so we can't us it. Need to come up with an alternative
    * rate to use. The process goes like this:
    * * If there are no rates to pick from, we're stuck and return 0 (error)
    * * If there are some rates, we pick the next one higher than the requested
    *   rate to use.
    * * If there aren't any higher, we use the highest available rate */

   if (rates.empty()) {
      /* we're stuck - there are no supported rates with this hardware. Error */
      wxLogDebug(wxT("GetBestRate() Error - no supported sample rates"));
      retval = 0.0;
      goto finished;
   }
   int i;
   for (i = 0; i < (int)rates.size(); i++)  // for each supported rate
         {
         if (rates[i] > rate) {
            // supported rate is greater than requested rate
            wxLogDebug(wxT("GetBestRate() Returning next higher rate - %.0ld Hz"), rates[i]);
            retval = rates[i];
            goto finished;
         }
         }

   wxLogDebug(wxT("GetBestRate() Returning highest rate - %.0ld Hz"), rates.back());
   retval = rates.back(); // the highest available rate
   goto finished;

finished:
   mCachedBestRateIn = sampleRate;
   mCachedBestRateOut = retval;
   mCachedBestRatePlaying = playing;
   mCachedBestRateCapturing = capturing;
   return retval;
}

double AudioIO::GetStreamTime()
{
   // Track time readout for the main thread

   if( !IsStreamActive() )
      return BAD_STREAM_TIME;

   return mPlaybackSchedule.GetTrackTime();
}


//////////////////////////////////////////////////////////////////////
//
//     Audio Thread Context
//
//////////////////////////////////////////////////////////////////////

AudioThread::ExitCode AudioThread::Entry()
{
   AudioIO *gAudioIO;
   while( !TestDestroy() &&
      nullptr != ( gAudioIO = AudioIO::Get() ) )
   {
      using Clock = std::chrono::steady_clock;
      auto loopPassStart = Clock::now();
      auto &schedule = gAudioIO->mPlaybackSchedule;
      const auto interval = schedule.GetPolicy().SleepInterval(schedule);

      // Set LoopActive outside the tests to avoid race condition
      gAudioIO->mAudioThreadTrackBufferExchangeLoopActive = true;
      if( gAudioIO->mAudioThreadShouldCallTrackBufferExchangeOnce )
      {
         gAudioIO->TrackBufferExchange();
         gAudioIO->mAudioThreadShouldCallTrackBufferExchangeOnce = false;
      }
      else if( gAudioIO->mAudioThreadTrackBufferExchangeLoopRunning )
      {
         gAudioIO->TrackBufferExchange();
      }
      gAudioIO->mAudioThreadTrackBufferExchangeLoopActive = false;

      std::this_thread::sleep_until( loopPassStart + interval );
   }

   return 0;
}


size_t AudioIO::GetCommonlyFreePlayback()
{
   auto commonlyAvail = mPlaybackBuffers[0]->AvailForPut();
   for (unsigned i = 1; i < mPlaybackTracks.size(); ++i)
      commonlyAvail = std::min(commonlyAvail,
         mPlaybackBuffers[i]->AvailForPut());
   // MB: subtract a few samples because the code in TrackBufferExchange has rounding
   // errors
   return commonlyAvail - std::min(size_t(10), commonlyAvail);
}

size_t AudioIoCallback::GetCommonlyReadyPlayback()
{
   auto commonlyAvail = mPlaybackBuffers[0]->AvailForGet();
   for (unsigned i = 1; i < mPlaybackTracks.size(); ++i)
      commonlyAvail = std::min(commonlyAvail,
         mPlaybackBuffers[i]->AvailForGet());
   return commonlyAvail;
}

size_t AudioIO::GetCommonlyAvailCapture()
{
   auto commonlyAvail = mCaptureBuffers[0]->AvailForGet();
   for (unsigned i = 1; i < mCaptureTracks.size(); ++i)
      commonlyAvail = std::min(commonlyAvail,
         mCaptureBuffers[i]->AvailForGet());
   return commonlyAvail;
}

// This method is the data gateway between the audio thread (which
// communicates with the disk) and the PortAudio callback thread
// (which communicates with the audio device).
void AudioIO::TrackBufferExchange()
{
   FillPlayBuffers();
   DrainRecordBuffers();
}

void AudioIO::FillPlayBuffers()
{
   if (mNumPlaybackChannels == 0)
      return;

   // Though extremely unlikely, it is possible that some buffers
   // will have more samples available than others.  This could happen
   // if we hit this code during the PortAudio callback.  To keep
   // things simple, we only write as much data as is vacant in
   // ALL buffers, and advance the global time by that much.
   auto nAvailable = GetCommonlyFreePlayback();

   // Don't fill the buffers at all unless we can do the
   // full mMaxPlaybackSecsToCopy.  This improves performance
   // by not always trying to process tiny chunks, eating the
   // CPU unnecessarily.
   if (nAvailable < mPlaybackSamplesToCopy)
      return;

   auto &policy = mPlaybackSchedule.GetPolicy();

   // More than mPlaybackSamplesToCopy might be copied:
   // May produce a larger amount when initially priming the buffer, or
   // perhaps again later in play to avoid underfilling the queue and falling
   // behind the real-time demand on the consumer side in the callback.
   auto nReady = GetCommonlyReadyPlayback();
   auto nNeeded =
      mPlaybackQueueMinimum - std::min(mPlaybackQueueMinimum, nReady);

   // wxASSERT( nNeeded <= nAvailable );

   // Limit maximum buffer size (increases performance)
   auto available = std::min( nAvailable,
      std::max( nNeeded, mPlaybackSamplesToCopy ) );

   // msmeyer: When playing a very short selection in looped
   // mode, the selection must be copied to the buffer multiple
   // times, to ensure, that the buffer has a reasonable size
   // This is the purpose of this loop.
   // PRL: or, when scrubbing, we may get work repeatedly from the
   // user interface.
   bool done = false;
   do {
      const auto [frames, toProduce] =
         policy.GetPlaybackSlice(mPlaybackSchedule, available);

      // Update the time queue.  This must be done before writing to the
      // ring buffers of samples, for proper synchronization with the
      // consumer side in the PortAudio thread, which reads the time
      // queue after reading the sample queues.  The sample queues use
      // atomic variables, the time queue doesn't.
      mPlaybackSchedule.mTimeQueue.Producer(mPlaybackSchedule, frames);

      for (size_t i = 0; i < mPlaybackTracks.size(); i++)
      {
         // The mixer here isn't actually mixing: it's just doing
         // resampling, format conversion, and possibly time track
         // warping
         if (frames > 0)
         {
            size_t produced = 0;
            if ( toProduce )
               produced = mPlaybackMixers[i]->Process( toProduce );
            //wxASSERT(produced <= toProduce);
            auto warpedSamples = mPlaybackMixers[i]->GetBuffer();
            const auto put = mPlaybackBuffers[i]->Put(
               warpedSamples, floatSample, produced, frames - produced);
            // wxASSERT(put == frames);
            // but we can't assert in this thread
            wxUnusedVar(put);
         }
      }

      if (mPlaybackTracks.empty())
         // Produce silence in the single ring buffer
         mPlaybackBuffers[0]->Put(nullptr, floatSample, 0, frames);

      available -= frames;
      // wxASSERT(available >= 0); // don't assert on this thread

      done = policy.RepositionPlayback( mPlaybackSchedule, mPlaybackMixers,
         frames, available );
   } while (available && !done);
}

void AudioIO::DrainRecordBuffers()
{
   if (mRecordingException || mCaptureTracks.empty())
      return;

   auto delayedHandler = [this] ( AudacityException * pException ) {
      // In the main thread, stop recording
      // This is one place where the application handles disk
      // exhaustion exceptions from wave track operations, without rolling
      // back to the last pushed undo state.  Instead, partial recording
      // results are pushed as a NEW undo state.  For this reason, as
      // commented elsewhere, we want an exception safety guarantee for
      // the output wave tracks, after the failed append operation, that
      // the tracks remain as they were after the previous successful
      // (block-level) appends.

      // Note that the Flush in StopStream() may throw another exception,
      // but StopStream() contains that exception, and the logic in
      // AudacityException::DelayedHandlerAction prevents redundant message
      // boxes.
      StopStream();
      DefaultDelayedHandlerAction{}( pException );
   };

   GuardedCall( [&] {
      // start record buffering
      const auto avail = GetCommonlyAvailCapture(); // samples
      const auto remainingTime =
         std::max(0.0, mRecordingSchedule.ToConsume());
      // This may be a very big double number:
      const auto remainingSamples = remainingTime * mRate;
      bool latencyCorrected = true;

      double deltat = avail / mRate;

      if (mAudioThreadShouldCallTrackBufferExchangeOnce ||
          deltat >= mMinCaptureSecsToCopy)
      {
         // This scope may combine many appendings of wave tracks,
         // and also an autosave, into one transaction,
         // lessening the number of checkpoints
         Optional<TransactionScope> pScope;
         auto pOwningProject = mOwningProject.lock();
         if (pOwningProject) {
            auto &pIO = ProjectFileIO::Get( *pOwningProject );
            pScope.emplace(pIO.GetConnection(), "Recording");
         }

         bool newBlocks = false;

         // Append captured samples to the end of the WaveTracks.
         // The WaveTracks have their own buffering for efficiency.
         auto numChannels = mCaptureTracks.size();

         for( size_t i = 0; i < numChannels; i++ )
         {
            sampleFormat trackFormat = mCaptureTracks[i]->GetSampleFormat();

            size_t discarded = 0;

            if (!mRecordingSchedule.mLatencyCorrected) {
               const auto correction = mRecordingSchedule.TotalCorrection();
               if (correction >= 0) {
                  // Rightward shift
                  // Once only (per track per recording), insert some initial
                  // silence.
                  size_t size = floor( correction * mRate * mFactor);
                  SampleBuffer temp(size, trackFormat);
                  ClearSamples(temp.ptr(), trackFormat, 0, size);
                  mCaptureTracks[i]->Append(temp.ptr(), trackFormat, size, 1);
               }
               else {
                  // Leftward shift
                  // discard some samples from the ring buffers.
                  size_t size = floor(
                     mRecordingSchedule.ToDiscard() * mRate );

                  // The ring buffer might have grown concurrently -- don't discard more
                  // than the "avail" value noted above.
                  discarded = mCaptureBuffers[i]->Discard(std::min(avail, size));

                  if (discarded < size)
                     // We need to visit this again to complete the
                     // discarding.
                     latencyCorrected = false;
               }
            }

            const float *pCrossfadeSrc = nullptr;
            size_t crossfadeStart = 0, totalCrossfadeLength = 0;
            if (i < mRecordingSchedule.mCrossfadeData.size())
            {
               // Do crossfading
               // The supplied crossfade samples are at the same rate as the track
               const auto &data = mRecordingSchedule.mCrossfadeData[i];
               totalCrossfadeLength = data.size();
               if (totalCrossfadeLength) {
                  crossfadeStart =
                     floor(mRecordingSchedule.Consumed() * mCaptureTracks[i]->GetRate());
                  if (crossfadeStart < totalCrossfadeLength)
                     pCrossfadeSrc = data.data() + crossfadeStart;
               }
            }

            wxASSERT(discarded <= avail);
            size_t toGet = avail - discarded;
            SampleBuffer temp;
            size_t size;
            sampleFormat format;
            if( mFactor == 1.0 )
            {
               // Take captured samples directly
               size = toGet;
               if (pCrossfadeSrc)
                  // Change to float for crossfade calculation
                  format = floatSample;
               else
                  format = trackFormat;
               temp.Allocate(size, format);
               const auto got =
                  mCaptureBuffers[i]->Get(temp.ptr(), format, toGet);
               // wxASSERT(got == toGet);
               // but we can't assert in this thread
               wxUnusedVar(got);
               if (double(size) > remainingSamples)
                  size = floor(remainingSamples);
            }
            else
            {
               size = lrint(toGet * mFactor);
               format = floatSample;
               SampleBuffer temp1(toGet, floatSample);
               temp.Allocate(size, format);
               const auto got =
                  mCaptureBuffers[i]->Get(temp1.ptr(), floatSample, toGet);
               // wxASSERT(got == toGet);
               // but we can't assert in this thread
               wxUnusedVar(got);
               /* we are re-sampling on the fly. The last resampling call
                * must flush any samples left in the rate conversion buffer
                * so that they get recorded
                */
               if (toGet > 0 ) {
                  if (double(toGet) > remainingSamples)
                     toGet = floor(remainingSamples);
                  const auto results =
                  mResample[i]->Process(mFactor, (float *)temp1.ptr(), toGet,
                                        !IsStreamActive(), (float *)temp.ptr(), size);
                  size = results.second;
               }
            }

            if (pCrossfadeSrc) {
               wxASSERT(format == floatSample);
               size_t crossfadeLength = std::min(size, totalCrossfadeLength - crossfadeStart);
               if (crossfadeLength) {
                  auto ratio = double(crossfadeStart) / totalCrossfadeLength;
                  auto ratioStep = 1.0 / totalCrossfadeLength;
                  auto pCrossfadeDst = (float*)temp.ptr();

                  // Crossfade loop here
                  for (size_t ii = 0; ii < crossfadeLength; ++ii) {
                     *pCrossfadeDst = ratio * *pCrossfadeDst + (1.0 - ratio) * *pCrossfadeSrc;
                     ++pCrossfadeSrc, ++pCrossfadeDst;
                     ratio += ratioStep;
                  }
               }
            }

            // Now append
            // see comment in second handler about guarantee
            newBlocks = mCaptureTracks[i]->Append(temp.ptr(), format, size, 1)
               || newBlocks;
         } // end loop over capture channels

         // Now update the recording schedule position
         mRecordingSchedule.mPosition += avail / mRate;
         mRecordingSchedule.mLatencyCorrected = latencyCorrected;

         auto pListener = GetListener();
         if (pListener && newBlocks)
            pListener->OnAudioIONewBlocks(&mCaptureTracks);

         if (pScope)
            pScope->Commit();
      }
      // end of record buffering
   },
   // handler
   [this] ( AudacityException *pException ) {
      if ( pException ) {
         // So that we don't attempt to fill the recording buffer again
         // before the main thread stops recording
         SetRecordingException();
         return ;
      }
      else
         // Don't want to intercept other exceptions (?)
         throw;
   },
   delayedHandler );
}

void AudioIoCallback::SetListener(
   const std::shared_ptr< AudioIOListener > &listener)
{
   if (IsBusy())
      return;

   mListener = listener;
}

// Automated Input Level Adjustment - Automatically tries to find an acceptable input volume
#ifdef EXPERIMENTAL_AUTOMATED_INPUT_LEVEL_ADJUSTMENT

#include "ProjectStatus.h"

void AudioIO::AILAInitialize() {
   gPrefs->Read(wxT("/AudioIO/AutomatedInputLevelAdjustment"), &mAILAActive,         false);
   gPrefs->Read(wxT("/AudioIO/TargetPeak"),            &mAILAGoalPoint,      AILA_DEF_TARGET_PEAK);
   gPrefs->Read(wxT("/AudioIO/DeltaPeakVolume"),       &mAILAGoalDelta,      AILA_DEF_DELTA_PEAK);
   gPrefs->Read(wxT("/AudioIO/AnalysisTime"),          &mAILAAnalysisTime,   AILA_DEF_ANALYSIS_TIME);
   gPrefs->Read(wxT("/AudioIO/NumberAnalysis"),        &mAILATotalAnalysis,  AILA_DEF_NUMBER_ANALYSIS);
   mAILAGoalDelta         /= 100.0;
   mAILAGoalPoint         /= 100.0;
   mAILAAnalysisTime      /= 1000.0;
   mAILAMax                = 0.0;
   mAILALastStartTime      = max(0.0, mPlaybackSchedule.mT0);
   mAILAClipped            = false;
   mAILAAnalysisCounter    = 0;
   mAILAChangeFactor       = 1.0;
   mAILALastChangeType     = 0;
   mAILATopLevel           = 1.0;
   mAILAAnalysisEndTime    = -1.0;
}

void AudioIO::AILADisable() {
   mAILAActive = false;
}

bool AudioIO::AILAIsActive() {
   return mAILAActive;
}

void AudioIO::AILASetStartTime() {
   mAILAAbsolutStartTime = Pa_GetStreamTime(mPortStreamV19);
   wxPrintf("START TIME %f\n\n", mAILAAbsolutStartTime);
}

double AudioIO::AILAGetLastDecisionTime() {
   return mAILAAnalysisEndTime;
}

void AudioIO::AILAProcess(double maxPeak) {
   const auto proj = mOwningProject.lock();
   if (proj && mAILAActive) {
      if (mInputMeter && mInputMeter->IsClipping()) {
         mAILAClipped = true;
         wxPrintf("clipped");
      }

      mAILAMax = max(mAILAMax, maxPeak);

      if ((mAILATotalAnalysis == 0 || mAILAAnalysisCounter < mAILATotalAnalysis) && mPlaybackSchedule.GetTrackTime() - mAILALastStartTime >= mAILAAnalysisTime) {
         auto ToLinearIfDB = [](double value, int dbRange) {
            if (dbRange >= 0)
               value = pow(10.0, (-(1.0-value) * dbRange)/20.0);
            return value;
         };

         putchar('\n');
         mAILAMax = mInputMeter ? ToLinearIfDB(mAILAMax, mInputMeter->GetDBRange()) : 0.0;
         double iv = (double) Px_GetInputVolume(mPortMixer);
         unsigned short changetype = 0; //0 - no change, 1 - increase change, 2 - decrease change
         wxPrintf("mAILAAnalysisCounter:%d\n", mAILAAnalysisCounter);
         wxPrintf("\tmAILAClipped:%d\n", mAILAClipped);
         wxPrintf("\tmAILAMax (linear):%f\n", mAILAMax);
         wxPrintf("\tmAILAGoalPoint:%f\n", mAILAGoalPoint);
         wxPrintf("\tmAILAGoalDelta:%f\n", mAILAGoalDelta);
         wxPrintf("\tiv:%f\n", iv);
         wxPrintf("\tmAILAChangeFactor:%f\n", mAILAChangeFactor);
         if (mAILAClipped || mAILAMax > mAILAGoalPoint + mAILAGoalDelta) {
            wxPrintf("too high:\n");
            mAILATopLevel = min(mAILATopLevel, iv);
            wxPrintf("\tmAILATopLevel:%f\n", mAILATopLevel);
            //if clipped or too high
            if (iv <= LOWER_BOUND) {
               //we can't improve it more now
               if (mAILATotalAnalysis != 0) {
                  mAILAActive = false;
                  ProjectStatus::Get( *proj ).Set(
                     XO(
"Automated Recording Level Adjustment stopped. It was not possible to optimize it more. Still too high.") );
               }
               wxPrintf("\talready min vol:%f\n", iv);
            }
            else {
               float vol = (float) max(LOWER_BOUND, iv+(mAILAGoalPoint-mAILAMax)*mAILAChangeFactor);
               Px_SetInputVolume(mPortMixer, vol);
               auto msg = XO(
"Automated Recording Level Adjustment decreased the volume to %f.").Format( vol );
               ProjectStatus::Get( *proj ).Set(msg);
               changetype = 1;
               wxPrintf("\tnew vol:%f\n", vol);
               float check = Px_GetInputVolume(mPortMixer);
               wxPrintf("\tverified %f\n", check);
            }
         }
         else if ( mAILAMax < mAILAGoalPoint - mAILAGoalDelta ) {
            //if too low
            wxPrintf("too low:\n");
            if (iv >= UPPER_BOUND || iv + 0.005 > mAILATopLevel) { //condition for too low volumes and/or variable volumes that cause mAILATopLevel to decrease too much
               //we can't improve it more
               if (mAILATotalAnalysis != 0) {
                  mAILAActive = false;
                  ProjectStatus::Get( *proj ).Set(
                     XO(
"Automated Recording Level Adjustment stopped. It was not possible to optimize it more. Still too low.") );
               }
               wxPrintf("\talready max vol:%f\n", iv);
            }
            else {
               float vol = (float) min(UPPER_BOUND, iv+(mAILAGoalPoint-mAILAMax)*mAILAChangeFactor);
               if (vol > mAILATopLevel) {
                  vol = (iv + mAILATopLevel)/2.0;
                  wxPrintf("\tTruncated vol:%f\n", vol);
               }
               Px_SetInputVolume(mPortMixer, vol);
               auto msg = XO(
"Automated Recording Level Adjustment increased the volume to %.2f.")
                  .Format( vol );
               ProjectStatus::Get( *proj ).Set(msg);
               changetype = 2;
               wxPrintf("\tnew vol:%f\n", vol);
               float check = Px_GetInputVolume(mPortMixer);
               wxPrintf("\tverified %f\n", check);
            }
         }

         mAILAAnalysisCounter++;
         //const PaStreamInfo* info = Pa_GetStreamInfo(mPortStreamV19);
         //double latency = 0.0;
         //if (info)
         //   latency = info->inputLatency;
         //mAILAAnalysisEndTime = mTime+latency;
         mAILAAnalysisEndTime = Pa_GetStreamTime(mPortStreamV19) - mAILAAbsolutStartTime;
         mAILAMax             = 0;
         wxPrintf("\tA decision was made @ %f\n", mAILAAnalysisEndTime);
         mAILAClipped         = false;
         mAILALastStartTime   = mPlaybackSchedule.GetTrackTime();

         if (changetype == 0)
            mAILAChangeFactor *= 0.8; //time factor
         else if (mAILALastChangeType == changetype)
            mAILAChangeFactor *= 1.1; //concordance factor
         else
            mAILAChangeFactor *= 0.7; //discordance factor
         mAILALastChangeType = changetype;
         putchar('\n');
      }

      if (mAILAActive && mAILATotalAnalysis != 0 && mAILAAnalysisCounter >= mAILATotalAnalysis) {
         mAILAActive = false;
         if (mAILAMax > mAILAGoalPoint + mAILAGoalDelta)
            ProjectStatus::Get( *proj ).Set(
               XO(
"Automated Recording Level Adjustment stopped. The total number of analyses has been exceeded without finding an acceptable volume. Still too high.") );
         else if (mAILAMax < mAILAGoalPoint - mAILAGoalDelta)
            ProjectStatus::Get( *proj ).Set(
               XO(
"Automated Recording Level Adjustment stopped. The total number of analyses has been exceeded without finding an acceptable volume. Still too low.") );
         else {
            auto msg = XO(
"Automated Recording Level Adjustment stopped. %.2f seems an acceptable volume.")
               .Format( Px_GetInputVolume(mPortMixer) );
            ProjectStatus::Get( *proj ).Set(msg);
         }
      }
   }
}
#endif

#define MAX(a,b) ((a) > (b) ? (a) : (b))

static void DoSoftwarePlaythrough(constSamplePtr inputBuffer,
                                  sampleFormat inputFormat,
                                  unsigned inputChannels,
                                  float *outputBuffer,
                                  unsigned long len)
{
   for (unsigned int i=0; i < inputChannels; i++) {
      auto inputPtr = inputBuffer + (i * SAMPLE_SIZE(inputFormat));

      SamplesToFloats(inputPtr, inputFormat,
         outputBuffer + i, len, inputChannels, 2);
   }

   // One mono input channel goes to both output channels...
   if (inputChannels == 1)
      for (int i=0; i < len; i++)
         outputBuffer[2*i + 1] = outputBuffer[2*i];
}

int audacityAudioCallback(const void *inputBuffer, void *outputBuffer,
                          unsigned long framesPerBuffer,
                          const PaStreamCallbackTimeInfo *timeInfo,
                          const PaStreamCallbackFlags statusFlags, void *userData )
{
   auto gAudioIO = AudioIO::Get();
   return gAudioIO->AudioCallback(
      static_cast<constSamplePtr>(inputBuffer),
      static_cast<float*>(outputBuffer), framesPerBuffer,
      timeInfo, statusFlags, userData);
}

// Stop recording if 'silence' is detected
// Start recording if sound detected.
//
//   By using CallAfter(), we can schedule the call to the toolbar
//   to run in the main GUI thread after the next event loop iteration.
//   That's important, because Pause() updates GUI, such as status bar,
//   and that should NOT happen in this audio non-gui thread.
void AudioIoCallback::CheckSoundActivatedRecordingLevel(
      float *inputSamples,
      unsigned long framesPerBuffer
   )
{
   // Quick returns if next to nothing to do.
   if( !mPauseRec )
      return;

   float maxPeak = 0.;
   for( unsigned long i = 0, cnt = framesPerBuffer * mNumCaptureChannels; i < cnt; ++i ) {
      float sample = fabs(*(inputSamples++));
      if (sample > maxPeak) {
         maxPeak = sample;
      }
   }

   bool bShouldBePaused = maxPeak < mSilenceLevel;
   if( bShouldBePaused != IsPaused() )
   {
      auto pListener = GetListener();
      if ( pListener )
         pListener->OnSoundActivationThreshold();
   }
}

// A function to apply the requested gain, fading up or down from the
// most recently applied gain.
void AudioIoCallback::AddToOutputChannel( unsigned int chan,
   float * outputMeterFloats,
   float * outputFloats,
   const float * tempBuf,
   bool drop,
   unsigned long len,
   WaveTrack *vt
   )
{
   const auto numPlaybackChannels = mNumPlaybackChannels;

   float gain = vt->GetChannelGain(chan);
   if (drop || mForceFadeOut.load(std::memory_order_relaxed) || mPaused)
      gain = 0.0;

   // Output volume emulation: possibly copy meter samples, then
   // apply volume, then copy to the output buffer
   if (outputMeterFloats != outputFloats)
      for ( unsigned i = 0; i < len; ++i)
         outputMeterFloats[numPlaybackChannels*i+chan] +=
            gain*tempBuf[i];

   // DV: We use gain to emulate panning.
   // Let's keep the old behavior for panning.
   gain *= ExpGain(mMixerOutputVol);

   float oldGain = vt->GetOldChannelGain(chan);
   if( gain != oldGain )
      vt->SetOldChannelGain(chan, gain);
   // if no microfades, jump in volume.
   if( !mbMicroFades )
      oldGain =gain;
   wxASSERT(len > 0);

   // Linear interpolate.
   float deltaGain = (gain - oldGain) / len;
   for (unsigned i = 0; i < len; i++)
      outputFloats[numPlaybackChannels*i+chan] += (oldGain + deltaGain * i) *tempBuf[i];
};

// Limit values to -1.0..+1.0
void ClampBuffer(float * pBuffer, unsigned long len){
   for(unsigned i = 0; i < len; i++)
      pBuffer[i] = wxClip( pBuffer[i], -1.0f, 1.0f);
};


// return true, IFF we have fully handled the callback.
//
// Mix and copy to PortAudio's output buffer
// from our intermediate playback buffers
//
bool AudioIoCallback::FillOutputBuffers(
   float *outputBuffer,
   unsigned long framesPerBuffer,
   float *outputMeterFloats
)
{
   const auto numPlaybackTracks = mPlaybackTracks.size();
   const auto numPlaybackChannels = mNumPlaybackChannels;
   const auto numCaptureChannels = mNumCaptureChannels;

   mMaxFramesOutput = 0;

   // Quick returns if next to nothing to do.
   if (mStreamToken <= 0 ||
       !outputBuffer ||
       numPlaybackChannels <= 0) {
      // So that UpdateTimePosition() will be correct, in case of MIDI play with
      // no audio output channels
      mMaxFramesOutput = framesPerBuffer;
      return false;
   }

   float *outputFloats = outputBuffer;

   if (mSeek && !mPlaybackSchedule.GetPolicy().AllowSeek(mPlaybackSchedule))
      mSeek = 0.0;

   if (mSeek){
      mCallbackReturn = CallbackDoSeek();
      return true;
   }

   // ------ MEMORY ALLOCATION ----------------------
   // These are small structures.
   WaveTrack **chans = (WaveTrack **) alloca(numPlaybackChannels * sizeof(WaveTrack *));
   float **tempBufs = (float **) alloca(numPlaybackChannels * sizeof(float *));

   // And these are larger structures....
   for (unsigned int c = 0; c < numPlaybackChannels; c++)
      tempBufs[c] = (float *) alloca(framesPerBuffer * sizeof(float));
   // ------ End of MEMORY ALLOCATION ---------------

   auto & em = RealtimeEffectManager::Get();
   em.RealtimeProcessStart();

   bool selected = false;
   int group = 0;
   int chanCnt = 0;

   // Choose a common size to take from all ring buffers
   const auto toGet =
      std::min<size_t>(framesPerBuffer, GetCommonlyReadyPlayback());

   // The drop and dropQuickly booleans are so named for historical reasons.
   // JKC: The original code attempted to be faster by doing nothing on silenced audio.
   // This, IMHO, is 'premature optimisation'.  Instead clearer and cleaner code would
   // simply use a gain of 0.0 for silent audio and go on through to the stage of
   // applying that 0.0 gain to the data mixed into the buffer.
   // Then (and only then) we would have if needed fast paths for:
   // - Applying a uniform gain of 0.0.
   // - Applying a uniform gain of 1.0.
   // - Applying some other uniform gain.
   // - Applying a linearly interpolated gain.
   // I would expect us not to need the fast paths, since linearly interpolated gain
   // is very cheap to process.

   bool drop = false;        // Track should become silent.
   bool dropQuickly = false; // Track has already been faded to silence.
   for (unsigned t = 0; t < numPlaybackTracks; t++)
   {
      WaveTrack *vt = mPlaybackTracks[t].get();
      chans[chanCnt] = vt;

      // TODO: more-than-two-channels
      auto nextTrack =
         t + 1 < numPlaybackTracks
            ? mPlaybackTracks[t + 1].get()
            : nullptr;

      // First and last channel in this group (for example left and right
      // channels of stereo).
      bool firstChannel = vt->IsLeader();
      bool lastChannel = !nextTrack || nextTrack->IsLeader();

      if ( firstChannel )
      {
         selected = vt->GetSelected();
         // IF mono THEN clear 'the other' channel.
         if ( lastChannel && (numPlaybackChannels>1)) {
            // TODO: more-than-two-channels
            memset(tempBufs[1], 0, framesPerBuffer * sizeof(float));
         }
         drop = TrackShouldBeSilent( *vt );
         dropQuickly = drop;
      }

      if( mbMicroFades )
         dropQuickly = dropQuickly && TrackHasBeenFadedOut( *vt );

      decltype(framesPerBuffer) len = 0;

      if (dropQuickly)
      {
         len = mPlaybackBuffers[t]->Discard(toGet);
         // keep going here.
         // we may still need to issue a paComplete.
      }
      else
      {
         len = mPlaybackBuffers[t]->Get((samplePtr)tempBufs[chanCnt],
                                                   floatSample,
                                                   toGet);
         // wxASSERT( len == toGet );
         if (len < framesPerBuffer)
            // This used to happen normally at the end of non-looping
            // plays, but it can also be an anomalous case where the
            // supply from TrackBufferExchange fails to keep up with the
            // real-time demand in this thread (see bug 1932).  We
            // must supply something to the sound card, so pad it with
            // zeroes and not random garbage.
            memset((void*)&tempBufs[chanCnt][len], 0,
               (framesPerBuffer - len) * sizeof(float));
         chanCnt++;
      }

      // PRL:  Bug1104:
      // There can be a difference of len in different loop passes if one channel
      // of a stereo track ends before the other!  Take a max!

      // PRL:  More recent rewrites of TrackBufferExchange should guarantee a
      // padding out of the ring buffers so that equal lengths are
      // available, so maxLen ought to increase from 0 only once
      mMaxFramesOutput = std::max(mMaxFramesOutput, len);

      if ( !lastChannel )
         continue;

      // Last channel of a track seen now
      len = mMaxFramesOutput;

      if( !dropQuickly && selected )
         len = em.RealtimeProcess(group, chanCnt, tempBufs, len);
      group++;

      CallbackCheckCompletion(mCallbackReturn, len);
      if (dropQuickly) // no samples to process, they've been discarded
         continue;

      // Our channels aren't silent.  We need to pass their data on.
      //
      // Note that there are two kinds of channel count.
      // c and chanCnt are counting channels in the Tracks.
      // chan (and numPlayBackChannels) is counting output channels on the device.
      // chan = 0 is left channel
      // chan = 1 is right channel.
      //
      // Each channel in the tracks can output to more than one channel on the device.
      // For example mono channels output to both left and right output channels.
      if (len > 0) for (int c = 0; c < chanCnt; c++)
      {
         vt = chans[c];

         if (vt->GetChannelIgnoringPan() == Track::LeftChannel ||
               vt->GetChannelIgnoringPan() == Track::MonoChannel )
            AddToOutputChannel( 0, outputMeterFloats, outputFloats,
               tempBufs[c], drop, len, vt);

         if (vt->GetChannelIgnoringPan() == Track::RightChannel ||
               vt->GetChannelIgnoringPan() == Track::MonoChannel  )
            AddToOutputChannel( 1, outputMeterFloats, outputFloats,
               tempBufs[c], drop, len, vt);
      }

      chanCnt = 0;
   }

   // Poke: If there are no playback tracks, then the earlier check
   // about the time indicator being past the end won't happen;
   // do it here instead (but not if looping or scrubbing)
   // PRL:  Also consume from the single playback ring buffer
   if (numPlaybackTracks == 0) {
      mMaxFramesOutput = mPlaybackBuffers[0]->Discard(toGet);
      CallbackCheckCompletion(mCallbackReturn, 0);
   }

   // wxASSERT( maxLen == toGet );

   em.RealtimeProcessEnd();
   mLastPlaybackTimeMillis = ::wxGetUTCTimeMillis();

   ClampBuffer( outputFloats, framesPerBuffer*numPlaybackChannels );
   if (outputMeterFloats != outputFloats)
      ClampBuffer( outputMeterFloats, framesPerBuffer*numPlaybackChannels );

   return false;
}

void AudioIoCallback::UpdateTimePosition(unsigned long framesPerBuffer)
{
   // Quick returns if next to nothing to do.
   if (mStreamToken <= 0)
      return;

   // Update the position seen by drawing code
   mPlaybackSchedule.SetTrackTime(
      mPlaybackSchedule.mTimeQueue.Consumer( mMaxFramesOutput, mRate ) );
}

// return true, IFF we have fully handled the callback.
//
// Copy from PortAudio input buffers to our intermediate recording buffers.
//
void AudioIoCallback::DrainInputBuffers(
   constSamplePtr inputBuffer,
   unsigned long framesPerBuffer,
   const PaStreamCallbackFlags statusFlags,
   float * tempFloats
)
{
   const auto numPlaybackTracks = mPlaybackTracks.size();
   const auto numPlaybackChannels = mNumPlaybackChannels;
   const auto numCaptureChannels = mNumCaptureChannels;

   // Quick returns if next to nothing to do.
   if (mStreamToken <= 0)
      return;
   if( !inputBuffer )
      return;
   if( numCaptureChannels <= 0 )
      return;

   // If there are no playback tracks, and we are recording, then the
   // earlier checks for being past the end won't happen, so do it here.
   if (mPlaybackSchedule.GetPolicy().Done(mPlaybackSchedule, 0)) {
      mCallbackReturn = paComplete;
   }

   // The error likely from a too-busy CPU falling behind real-time data
   // is paInputOverflow
   bool inputError =
      (statusFlags & (paInputOverflow))
      && !(statusFlags & paPrimingOutput);

   // But it seems it's easy to get false positives, at least on Mac
   // So we have not decided to enable this extra detection yet in
   // production

   size_t len = framesPerBuffer;
   for(unsigned t = 0; t < numCaptureChannels; t++)
      len = std::min( len, mCaptureBuffers[t]->AvailForPut() );

   if (mSimulateRecordingErrors && 100LL * rand() < RAND_MAX)
      // Make spurious errors for purposes of testing the error
      // reporting
      len = 0;

   // A different symptom is that len < framesPerBuffer because
   // the other thread, executing TrackBufferExchange, isn't consuming fast
   // enough from mCaptureBuffers; maybe it's CPU-bound, or maybe the
   // storage device it writes is too slow
   if (mDetectDropouts &&
         ((mDetectUpstreamDropouts && inputError) ||
         len < framesPerBuffer) ) {
      // Assume that any good partial buffer should be written leftmost
      // and zeroes will be padded after; label the zeroes.
      auto start = mPlaybackSchedule.GetTrackTime() +
            len / mRate + mRecordingSchedule.mLatencyCorrection;
      auto duration = (framesPerBuffer - len) / mRate;
      auto pLast = mLostCaptureIntervals.empty()
         ? nullptr : &mLostCaptureIntervals.back();
      if (pLast &&
          fabs(pLast->first + pLast->second - start) < 0.5/mRate)
         // Make one bigger interval, not two butting intervals
         pLast->second = start + duration - pLast->first;
      else
         mLostCaptureIntervals.emplace_back( start, duration );
   }

   if (len < framesPerBuffer)
   {
      mLostSamples += (framesPerBuffer - len);
      wxPrintf(wxT("lost %d samples\n"), (int)(framesPerBuffer - len));
   }

   if (len <= 0)
      return;

   // We have an ASSERT in the AudioIO constructor to alert us to
   // possible issues with the (short*) cast.  We'd have a problem if
   // sizeof(short) > sizeof(float) since our buffers are sized for floats.
   for(unsigned t = 0; t < numCaptureChannels; t++) {

      // dmazzoni:
      // Un-interleave.  Ugly special-case code required because the
      // capture channels could be in three different sample formats;
      // it'd be nice to be able to call CopySamples, but it can't
      // handle multiplying by the gain and then clipping.  Bummer.

      switch(mCaptureFormat) {
         case floatSample: {
            auto inputFloats = (const float *)inputBuffer;
            for(unsigned i = 0; i < len; i++)
               tempFloats[i] =
                  inputFloats[numCaptureChannels*i+t];
         } break;
         case int24Sample:
            // We should never get here. Audacity's int24Sample format
            // is different from PortAudio's sample format and so we
            // make PortAudio return float samples when recording in
            // 24-bit samples.
            wxASSERT(false);
            break;
         case int16Sample: {
            auto inputShorts = (const short *)inputBuffer;
            short *tempShorts = (short *)tempFloats;
            for( unsigned i = 0; i < len; i++) {
               float tmp = inputShorts[numCaptureChannels*i+t];
               tmp = wxClip( -32768, tmp, 32767 );
               tempShorts[i] = (short)(tmp);
            }
         } break;
      } // switch

      // JKC: mCaptureFormat must be for samples with sizeof(float) or
      // fewer bytes (because tempFloats is sized for floats).  All
      // formats are 2 or 4 bytes, so we are OK.
      const auto put =
         mCaptureBuffers[t]->Put(
            (samplePtr)tempFloats, mCaptureFormat, len);
      // wxASSERT(put == len);
      // but we can't assert in this thread
      wxUnusedVar(put);
   }
}


#if 0
// Record the reported latency from PortAudio.
// TODO: Don't recalculate this with every callback?
// 01/21/2009:  Disabled until a better solution presents itself.
void OldCodeToCalculateLatency()
{
   // As of 06/17/2006, portaudio v19 returns inputBufferAdcTime set to
   // zero.  It is being worked on, but for now we just can't do much
   // but follow the leader.
   //
   // 08/27/2006: too inconsistent for now...just leave it a zero.
   //
   // 04/16/2008: Looks like si->inputLatency comes back with something useful though.
   // This rearranged logic uses si->inputLatency, but if PortAudio fixes inputBufferAdcTime,
   // this code won't have to be modified to use it.
   // Also avoids setting mLastRecordingOffset except when simultaneously playing and recording.
   //
   if (numCaptureChannels > 0 && numPlaybackChannels > 0) // simultaneously playing and recording
   {
      if (timeInfo->inputBufferAdcTime > 0)
         mLastRecordingOffset = timeInfo->inputBufferAdcTime - timeInfo->outputBufferDacTime;
      else if (mLastRecordingOffset == 0.0)
      {
         const PaStreamInfo* si = Pa_GetStreamInfo( mPortStreamV19 );
         mLastRecordingOffset = -si->inputLatency;
      }
   }
}
#endif


// return true, IFF we have fully handled the callback.
// Prime the output buffer with 0's, optionally adding in the playthrough.
void AudioIoCallback::DoPlaythrough(
      constSamplePtr inputBuffer,
      float *outputBuffer,
      unsigned long framesPerBuffer,
      float *outputMeterFloats
   )
{
   const auto numCaptureChannels = mNumCaptureChannels;
   const auto numPlaybackChannels = mNumPlaybackChannels;

   // Quick returns if next to nothing to do.
   if( !outputBuffer )
      return;
   if( numPlaybackChannels <= 0 )
      return;

   float *outputFloats = outputBuffer;
   for(unsigned i = 0; i < framesPerBuffer*numPlaybackChannels; i++)
      outputFloats[i] = 0.0;

   if (inputBuffer && mSoftwarePlaythrough) {
      DoSoftwarePlaythrough(inputBuffer, mCaptureFormat,
                              numCaptureChannels,
                              outputBuffer, framesPerBuffer);
   }

   // Copy the results to outputMeterFloats if necessary
   if (outputMeterFloats != outputFloats) {
      for (unsigned i = 0; i < framesPerBuffer*numPlaybackChannels; ++i) {
         outputMeterFloats[i] = outputFloats[i];
      }
   }
}

/* Send data to recording VU meter if applicable */
// Also computes rms
void AudioIoCallback::SendVuInputMeterData(
   const float *inputSamples,
   unsigned long framesPerBuffer
   )
{
   const auto numCaptureChannels = mNumCaptureChannels;

   auto pInputMeter = mInputMeter.lock();
   if ( !pInputMeter )
      return;
   if( pInputMeter->IsMeterDisabled())
      return;

   // get here if meters are actually live , and being updated
   /* It's critical that we don't update the meters while StopStream is
      * trying to stop PortAudio, otherwise it can lead to a freeze.  We use
      * two variables to synchronize:
      *   mUpdatingMeters tells StopStream when the callback is about to enter
      *     the code where it might update the meters, and
      *   mUpdateMeters is how the rest of the code tells the callback when it
      *     is allowed to actually do the updating.
      * Note that mUpdatingMeters must be set first to avoid a race condition.
      */
   //TODO use atomics instead.
   mUpdatingMeters = true;
   if (mUpdateMeters) {
         pInputMeter->UpdateDisplay(numCaptureChannels,
                                    framesPerBuffer,
                                    inputSamples);
   }
   mUpdatingMeters = false;
}

/* Send data to playback VU meter if applicable */
void AudioIoCallback::SendVuOutputMeterData(
   const float *outputMeterFloats,
   unsigned long framesPerBuffer)
{
   const auto numPlaybackChannels = mNumPlaybackChannels;

   auto pOutputMeter = mOutputMeter.lock();
   if (!pOutputMeter)
      return;
   if( pOutputMeter->IsMeterDisabled() )
      return;
   if( !outputMeterFloats)
      return;

   // Get here if playback meter is live
   /* It's critical that we don't update the meters while StopStream is
      * trying to stop PortAudio, otherwise it can lead to a freeze.  We use
      * two variables to synchronize:
      *  mUpdatingMeters tells StopStream when the callback is about to enter
      *    the code where it might update the meters, and
      *  mUpdateMeters is how the rest of the code tells the callback when it
      *    is allowed to actually do the updating.
      * Note that mUpdatingMeters must be set first to avoid a race condition.
      */
   mUpdatingMeters = true;
   if (mUpdateMeters) {
      pOutputMeter->UpdateDisplay(numPlaybackChannels,
                                             framesPerBuffer,
                                             outputMeterFloats);

      //v Vaughan, 2011-02-25: Moved this update back to TrackPanel::OnTimer()
      //    as it helps with playback issues reported by Bill and noted on Bug 258.
      //    The problem there occurs if Software Playthrough is on.
      //    Could conditionally do the update here if Software Playthrough is off,
      //    and in TrackPanel::OnTimer() if Software Playthrough is on, but not now.
      // PRL 12 Jul 2015: and what was in TrackPanel::OnTimer is now handled by means of event
      // type EVT_TRACK_PANEL_TIMER
      //MixerBoard* pMixerBoard = mOwningProject->GetMixerBoard();
      //if (pMixerBoard)
      //   pMixerBoard->UpdateMeters(GetStreamTime(),
      //                              (pProj->GetControlToolBar()->GetLastPlayMode() == loopedPlay));
   }
   mUpdatingMeters = false;
}

unsigned AudioIoCallback::CountSoloingTracks(){
   const auto numPlaybackTracks = mPlaybackTracks.size();

   // MOVE_TO: CountSoloedTracks() function
   unsigned numSolo = 0;
   for(unsigned t = 0; t < numPlaybackTracks; t++ )
      if( mPlaybackTracks[t]->GetSolo() )
         numSolo++;
   auto range = Extensions();
   numSolo += std::accumulate(range.begin(), range.end(), 0,
      [](unsigned sum, auto &ext){
         return sum + ext.CountOtherSoloTracks(); });
   return numSolo;
}

// TODO: Consider making the two Track status functions into functions of
// WaveTrack.

// true IFF the track should be silent.
// The track may not yet be silent, since it may still be
// fading out.
bool AudioIoCallback::TrackShouldBeSilent( const WaveTrack &wt )
{
   return mPaused || (!wt.GetSolo() && (
      // Cut if somebody else is soloing
      mbHasSoloTracks ||
      // Cut if we're muted (and not soloing)
      wt.GetMute()
   ));
}

// This is about micro-fades.
bool AudioIoCallback::TrackHasBeenFadedOut( const WaveTrack &wt )
{
   const auto channel = wt.GetChannelIgnoringPan();
   if ((channel == Track::LeftChannel  || channel == Track::MonoChannel) &&
      wt.GetOldChannelGain(0) != 0.0)
      return false;
   if ((channel == Track::RightChannel || channel == Track::MonoChannel) &&
      wt.GetOldChannelGain(1) != 0.0)
      return false;
   return true;
}

bool AudioIoCallback::AllTracksAlreadySilent()
{
   const bool dropAllQuickly = std::all_of(
      mPlaybackTracks.begin(), mPlaybackTracks.end(),
      [&]( const std::shared_ptr< WaveTrack > &vt )
         { return
      TrackShouldBeSilent( *vt ) &&
      TrackHasBeenFadedOut( *vt ); }
   );
   return dropAllQuickly;
}

AudioIoCallback::AudioIoCallback()
{
   auto &factories = AudioIOExt::GetFactories();
   for (auto &factory: factories)
      if (auto pExt = factory(mPlaybackSchedule))
         mAudioIOExt.push_back( move(pExt) );
}


AudioIoCallback::~AudioIoCallback()
{
}


int AudioIoCallback::AudioCallback(
   constSamplePtr inputBuffer, float *outputBuffer,
   unsigned long framesPerBuffer,
   const PaStreamCallbackTimeInfo *timeInfo,
   const PaStreamCallbackFlags statusFlags, void * WXUNUSED(userData) )
{
   // Poll tracks for change of state.  User might click mute and solo buttons.
   mbHasSoloTracks = CountSoloingTracks() > 0 ;
   mCallbackReturn = paContinue;

   if (IsPaused()
       // PRL:  Why was this added?  Was it only because of the mysterious
       // initial leading zeroes, now solved by setting mStreamToken early?
       // JKC: I think it's used for the MIDI time cursor.  See comments
       // at head of file about AudioTime().
       || mStreamToken <= 0
       )
      mNumPauseFrames += framesPerBuffer;

   for( auto &ext : Extensions() ) {
      ext.ComputeOtherTimings(mRate,
         timeInfo,
         framesPerBuffer);
      ext.FillOtherBuffers(
         mRate, mNumPauseFrames, IsPaused(), mbHasSoloTracks);
   }

   // ------ MEMORY ALLOCATIONS -----------------------------------------------
   // tempFloats will be a reusable scratch pad for (possibly format converted)
   // audio data.  One temporary use is for the InputMeter data.
   const auto numPlaybackChannels = mNumPlaybackChannels;
   const auto numCaptureChannels = mNumCaptureChannels;
   float *tempFloats = (float *)alloca(framesPerBuffer*sizeof(float)*
                             MAX(numCaptureChannels,numPlaybackChannels));

   bool bVolEmulationActive =
      (outputBuffer && mMixerOutputVol != 1.0);
   // outputMeterFloats is the scratch pad for the output meter.
   // we can often reuse the existing outputBuffer and save on allocating
   // something new.
   float *outputMeterFloats = bVolEmulationActive ?
         (float *)alloca(framesPerBuffer*numPlaybackChannels * sizeof(float)) :
         outputBuffer;
   // ----- END of MEMORY ALLOCATIONS ------------------------------------------

   if (inputBuffer && numCaptureChannels) {
      float *inputSamples;

      if (mCaptureFormat == floatSample) {
         inputSamples = (float *) inputBuffer;
      }
      else {
         SamplesToFloats(reinterpret_cast<constSamplePtr>(inputBuffer),
            mCaptureFormat, tempFloats, framesPerBuffer * numCaptureChannels);
         inputSamples = tempFloats;
      }

      SendVuInputMeterData(
         inputSamples,
         framesPerBuffer);

      // This function may queue up a pause or resume.
      // TODO this is a bit dodgy as it toggles the Pause, and
      // relies on an idle event to have handled that, so could
      // queue up multiple toggle requests and so do nothing.
      // Eventually it will sort itself out by random luck, but
      // the net effect is a delay in starting/stopping sound activated
      // recording.
      CheckSoundActivatedRecordingLevel(
         inputSamples,
         framesPerBuffer);
   }

   // Even when paused, we do playthrough.
   // Initialise output buffer to zero or to playthrough data.
   // Initialise output meter values.
   DoPlaythrough(
      inputBuffer,
      outputBuffer,
      framesPerBuffer,
      outputMeterFloats);

   // Test for no track audio to play (because we are paused and have faded out)
   if( mPaused &&  (( !mbMicroFades ) || AllTracksAlreadySilent() ))
      return mCallbackReturn;

   // To add track output to output (to play sound on speaker)
   // possible exit, if we were seeking.
   if( FillOutputBuffers(
         outputBuffer,
         framesPerBuffer,
         outputMeterFloats))
      return mCallbackReturn;

   // To move the cursor onwards.  (uses mMaxFramesOutput)
   UpdateTimePosition(framesPerBuffer);

   // To capture input into track (sound from microphone)
   DrainInputBuffers(
      inputBuffer,
      framesPerBuffer,
      statusFlags,
      tempFloats);

   SendVuOutputMeterData( outputMeterFloats, framesPerBuffer);

   return mCallbackReturn;
}

int AudioIoCallback::CallbackDoSeek()
{
   const int token = mStreamToken;
   wxMutexLocker locker(mSuspendAudioThread);
   if (token != mStreamToken)
      // This stream got destroyed while we waited for it
      return paAbort;

   const auto numPlaybackTracks = mPlaybackTracks.size();

   // Pause audio thread and wait for it to finish
   mAudioThreadTrackBufferExchangeLoopRunning = false;
   while( mAudioThreadTrackBufferExchangeLoopActive )
   {
      wxMilliSleep( 50 );
   }

   // Calculate the NEW time position, in the PortAudio callback
   const auto time =
      mPlaybackSchedule.GetPolicy().OffsetTrackTime( mPlaybackSchedule, mSeek );

   mPlaybackSchedule.SetTrackTime( time );
   mSeek = 0.0;


   // Reset mixer positions and flush buffers for all tracks
   for (size_t i = 0; i < numPlaybackTracks; i++)
   {
      const bool skipping = true;
      mPlaybackMixers[i]->Reposition( time, skipping );
      const auto toDiscard =
         mPlaybackBuffers[i]->AvailForGet();
      const auto discarded =
         mPlaybackBuffers[i]->Discard( toDiscard );
      // wxASSERT( discarded == toDiscard );
      // but we can't assert in this thread
      wxUnusedVar(discarded);
   }

   mPlaybackSchedule.mTimeQueue.Prime(time);

   // Reload the ring buffers
   mAudioThreadShouldCallTrackBufferExchangeOnce = true;
   while( mAudioThreadShouldCallTrackBufferExchangeOnce )
   {
      wxMilliSleep( 50 );
   }

   // Reenable the audio thread
   mAudioThreadTrackBufferExchangeLoopRunning = true;

   return paContinue;
}

void AudioIoCallback::CallbackCheckCompletion(
   int &callbackReturn, unsigned long len)
{
   if (mPaused)
      return;

   bool done =
      mPlaybackSchedule.GetPolicy().Done(mPlaybackSchedule, len);
   if (!done)
      return;

   for( auto &ext : Extensions() )
      ext.SignalOtherCompletion();
   callbackReturn = paComplete;
}

auto AudioIoCallback::AudioIOExtIterator::operator *() const -> AudioIOExt &
{
   // Down-cast and dereference are safe because only AudioIOCallback
   // populates the array
   return *static_cast<AudioIOExt*>(mIterator->get());
}

bool AudioIO::IsCapturing() const
{
   // Includes a test of mTime, used in the main thread
   return IsStreamActive() &&
      GetNumCaptureChannels() > 0 &&
      mPlaybackSchedule.GetTrackTime() >=
         mPlaybackSchedule.mT0 + mRecordingSchedule.mPreRoll;
}