AudioIO.cpp

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/**********************************************************************
 
  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.
 
********************************************************************//*****************************************************************//****************************************************************//****************************************************************//*******************************************************************/
#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>
#include <optional>
 
#ifdef __WXMSW__
#include <malloc.h>
#endif
 
#ifdef HAVE_ALLOCA_H
#include <alloca.h>
#endif
 
#include "portaudio.h"
#ifdef __WXMSW__
#include "pa_win_wasapi.h"
#endif
 
#if USE_PORTMIXER
#include "portmixer.h"
#endif
 
#include <wx/wxcrtvararg.h>
#include <wx/log.h>
#include <wx/time.h>
#include <wx/debug.h>
 
#if defined(__WXMAC__) || defined(__WXMSW__)
#include <wx/power.h>
#endif
 
#include "Channel.h"
#include "Meter.h"
#include "Mix.h"
#include "Resample.h"
#include "RingBuffer.h"
#include "Decibels.h"
#include "Prefs.h"
#include "Project.h"
#include "TransactionScope.h"
 
#include "RealtimeEffectManager.h"
#include "QualitySettings.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() );
}
 
struct AudioIoCallback::TransportState {
   TransportState(std::weak_ptr<AudacityProject> wOwningProject,
      const ConstPlayableSequences &playbackSequences,
      unsigned numPlaybackChannels, double sampleRate)
   {
      if (auto pOwningProject = wOwningProject.lock();
          pOwningProject && numPlaybackChannels > 0) {
         // Setup for realtime playback at the rate of the realtime
         // stream, not the rate of the sample sequence.
         mpRealtimeInitialization.emplace(
            move(wOwningProject), sampleRate, numPlaybackChannels);
         // The following adds a new effect processor for each logical sequence.
         for (size_t i = 0, cnt = playbackSequences.size(); i < cnt; ++i) {
            // An array only of non-null pointers should be given to us
            const auto vt = playbackSequences[i].get();
            const auto pGroup = vt ? vt->FindChannelGroup() : nullptr;
            if (!pGroup) {
               assert(false);
               continue;
            }
            mpRealtimeInitialization
               ->AddGroup(*pGroup, numPlaybackChannels, sampleRate);
         }
      }
   }
 
   std::optional<RealtimeEffects::InitializationScope> mpRealtimeInitialization;
};
 
// 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 );
 
 
//
//     UI Thread Context
//
 
void AudioIO::Init()
{
   auto pAudioIO = safenew AudioIO();
   ugAudioIO.reset(pAudioIO);
   pAudioIO->StartThread();
 
   // 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 ));
 
   mAudioThreadShouldCallSequenceBufferExchangeOnce
      .store(false, std::memory_order_relaxed);
   mAudioThreadSequenceBufferExchangeLoopRunning
      .store(false, std::memory_order_relaxed);
   mAudioThreadSequenceBufferExchangeLoopActive
      .store(false, std::memory_order_relaxed);
 
   mAudioThreadAcknowledge.store(Acknowledge::eNone, std::memory_order_relaxed);
 
   mPortStreamV19 = NULL;
 
   mNumPauseFrames = 0;
 
#ifdef EXPERIMENTAL_AUTOMATED_INPUT_LEVEL_ADJUSTMENT
   mAILAActive = false;
#endif
 
   mLastPaError = paNoError;
 
   mLastRecordingOffset = 0.0;
   mNumCaptureChannels = 0;
   mSilenceLevel = 0.0;
 
   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
      using namespace BasicUI;
      ShowMessageBox(
         errStr,
         MessageBoxOptions{}
            .Caption(XO("Error Initializing Audio"))
            .IconStyle(Icon::Error)
            .ButtonStyle(Button::Ok));
 
      // 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.
   }
 
#if defined(USE_PORTMIXER)
   mPortMixer = NULL;
   mPreviousHWPlaythrough = -1.0;
   HandleDeviceChange();
#else
   mInputMixerWorks = false;
#endif
 
   SetMixerOutputVol(AudioIOPlaybackVolume.Read());
 
   mLastPlaybackTimeMillis = 0;
}
 
void AudioIO::StartThread()
{
   mAudioThread = std::thread(AudioThread, ref(mFinishAudioThread));
}
 
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();
 
   mFinishAudioThread.store(true, std::memory_order_release);
   mAudioThread.join();
}
 
std::shared_ptr<RealtimeEffectState>
AudioIO::AddState(AudacityProject &project,
   ChannelGroup *pGroup, const PluginID & id)
{
   RealtimeEffects::InitializationScope *pInit = nullptr;
   if (mpTransportState && mpTransportState->mpRealtimeInitialization)
      if (auto pProject = GetOwningProject(); pProject.get() == &project)
         pInit = &*mpTransportState->mpRealtimeInitialization;
   return RealtimeEffectManager::Get(project).AddState(pInit, pGroup, id);
}
 
std::shared_ptr<RealtimeEffectState>
AudioIO::ReplaceState(AudacityProject &project,
   ChannelGroup *pGroup, size_t index, const PluginID & id)
{
   RealtimeEffects::InitializationScope *pInit = nullptr;
   if (mpTransportState && mpTransportState->mpRealtimeInitialization)
      if (auto pProject = GetOwningProject(); pProject.get() == &project)
         pInit = &*mpTransportState->mpRealtimeInitialization;
   return RealtimeEffectManager::Get(project)
      .ReplaceState(pInit, pGroup, index, id);
}
 
void AudioIO::RemoveState(AudacityProject &project,
   ChannelGroup *pGroup,
   const std::shared_ptr<RealtimeEffectState> pState)
{
   RealtimeEffects::InitializationScope *pInit = nullptr;
   if (mpTransportState && mpTransportState->mpRealtimeInitialization)
      if (auto pProject = GetOwningProject(); pProject.get() == &project)
         pInit = &*mpTransportState->mpRealtimeInitialization;
   RealtimeEffectManager::Get(project).RemoveState(pInit, pGroup, pState);
}
 
void AudioIO::SetMixer(int inputSource, float recordVolume,
                       float playbackVolume)
{
   SetMixerOutputVol(playbackVolume);
   AudioIOPlaybackVolume.Write(playbackVolume);
 
#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 = GetMixerOutputVol();
 
#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)
{
   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);
 
   // GetBestRate() will return 0.0 for bidirectional streams when there is no
   // common sample rate supported by both the input and output devices.
   // Bidirectional streams are used when recording while overdub or
   // software playthrough options are enabled.
 
   // Pa_OpenStream() will return paInvalidSampleRate when trying to create the
   // bidirectional stream with a sampleRate of 0.0
   bool isStreamBidirectional = (numCaptureChannels > 0) && (numPlaybackChannels > 0);
   bool isUnsupportedSampleRate = isStreamBidirectional && (mRate == 0.0);
 
   // July 2016 (Carsten and Uwe)
   // BUG 193: Tell PortAudio sound card will handle 24 bit (under DirectSound) using
   // userData.
   auto captureFormat = mCaptureFormat;
   auto 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;
      // mCaptureFormat is overwritten before the call to
      // StartPortAudioStream. So the hack with captureFormat_saved is
      // still working (assuming it worked before).
      mCaptureFormat = captureFormat;
   }
 
   mNumPlaybackChannels = numPlaybackChannels;
   mNumCaptureChannels = numCaptureChannels;
 
   bool usePlayback = false, useCapture = false;
   PaStreamParameters playbackParameters{};
   PaStreamParameters captureParameters{};
 
   #ifdef __WXMSW__
   PaWasapiStreamInfo wasapiStreamInfo{};
   #endif
 
   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;
 
      const PaHostApiInfo* hostInfo = Pa_GetHostApiInfo(playbackDeviceInfo->hostApi);
      bool isWASAPI = (hostInfo && hostInfo->type == paWASAPI);
 
      #ifdef __WXMSW__
      // If the host API is WASAPI, the stream is bidirectional and there is no
      // supported sample rate enable the WASAPI Sample Rate Conversion
      // for the playback device.
      if (isWASAPI && isUnsupportedSampleRate)
      {
         wasapiStreamInfo.size = sizeof(PaWasapiStreamInfo);
         wasapiStreamInfo.hostApiType = paWASAPI;
         wasapiStreamInfo.version = 1;
         wasapiStreamInfo.flags = paWinWasapiAutoConvert;
 
         playbackParameters.hostApiSpecificStreamInfo = &wasapiStreamInfo;
      }
      #endif
 
      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.
         playbackParameters.suggestedLatency = isWASAPI ? 0.0 : latencyDuration/1000.0;
      }
 
      mOutputMeter = options.playbackMeter;
   }
 
   if( numCaptureChannels > 0)
   {
      useCapture = true;
 
      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;
 
      const PaHostApiInfo* hostInfo = Pa_GetHostApiInfo(captureDeviceInfo->hostApi);
      bool isWASAPI = (hostInfo && hostInfo->type == paWASAPI);
 
      // If the stream is bidirectional and there is no supported sample rate
      // set mRate to the value supported by the capture device.
      if (isWASAPI && isUnsupportedSampleRate)
         mRate = captureDeviceInfo->defaultSampleRate;
 
      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 );
   }
 
   const auto deviceInfo = usePlayback ?
                              Pa_GetDeviceInfo(playbackParameters.device) :
                              Pa_GetDeviceInfo(captureParameters.device);
 
   if (deviceInfo != nullptr)
   {
      const auto hostApiInfo = Pa_GetHostApiInfo(deviceInfo->hostApi);
 
      if (hostApiInfo)
      {
         mUsingAlsa = hostApiInfo->type == paALSA;
         mUsingJack = hostApiInfo->type == paJACK;
      }
   }
 
   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__
   {
      using namespace std::chrono;
      if (DeviceManager::Instance()->GetTimeSinceRescan() < 10s)
         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) {
         const auto stream = Pa_GetStreamInfo(mPortStreamV19);
         // Use the reported latency as a hint about the hardware buffer size
         // required for uninterrupted playback.
         const auto outputLatency =
            mUsingJack ?
               // When using Jack as a host, PA calculates the wrong latency
               // if a non system port is used. Assume, that Jack provides a very
               // low latency, lower than user requested
               // (https://github.com/audacity/audacity/issues/4646)
               (latencyDuration / 1000.0) :
               // Otherwise, use the (likely incorrect) latency reported by PA
               stream->outputLatency;
 
         mHardwarePlaybackLatencyFrames = lrint(outputLatency * mRate);
#ifdef __WXGTK__
         // DV: When using ALSA PortAudio does not report the buffer size.
         // Instead, it reports periodSize * (periodsCount - 1). It is impossible
         // to retrieve periodSize or periodsCount either. By default PA sets
         // periodsCount to 4. However it was observed, that PA reports back ~100msec
         // latency and expects a buffer of ~200msecs on Audacity default settings
         // which suggests that ALSA changes the periodsCount to suit its needs.
         //
         // Why 3? 2 doesn't work for me, 3 does :-) So similar to PA - this
         // is the value that works for author setup.
         if (mUsingAlsa)
            mHardwarePlaybackLatencyFrames *= 3;
#endif
         break;
      }
      wxLogDebug("Attempt %u to open capture stream failed with: %d", 1 + tries, mLastPaError);
      using namespace std::chrono;
      std::this_thread::sleep_for(1s);
   }
 
 
#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;
   mCaptureFormat = captureFormat;
   mCaptureRate = 44100.0; // Shouldn't matter
   success = StartPortAudioStream(options,
      static_cast<unsigned int>(playbackChannels),
      static_cast<unsigned int>(captureChannels));
 
   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;
   }
 
   Publish({ pOwningProject.get(), AudioIOEvent::MONITOR, true });
 
   // 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 TransportSequences &sequences,
   double t0, double t1, double mixerLimit,
   const AudioIOStartStreamOptions &options)
{
   // precondition
   assert(std::all_of(
      sequences.playbackSequences.begin(), sequences.playbackSequences.end(),
      [](const auto &pSequence){
         const auto pGroup =
            pSequence ? pSequence->FindChannelGroup() : nullptr;
         return pGroup; }
   ));
 
   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) {
         using namespace std::chrono;
         std::this_thread::sleep_for(50ms);
      }
   }
 
   gPrefs->Read(wxT("/AudioIO/SWPlaythrough"), &mSoftwarePlaythrough, false);
   mPauseRec = SoundActivatedRecord.Read();
   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;
   mCaptureSequences = sequences.captureSequences;
   mPlaybackSequences = sequences.playbackSequences;
 
   bool commit = false;
   auto cleanupSequences = finally([&]{
      if (!commit) {
         // Don't keep unnecessary shared pointers to sequences
         mPlaybackSequences.clear();
         mCaptureSequences.clear();
         for(auto &ext : Extensions())
            ext.AbortOtherStream();
 
         // Don't cause a busy wait in the audio thread after stopping scrubbing
         mPlaybackSchedule.ResetMode();
      }
   });
 
   mPlaybackBuffers.clear();
   mScratchBuffers.clear();
   mScratchPointers.clear();
   mPlaybackMixers.clear();
   mCaptureBuffers.clear();
   mResample.clear();
   mPlaybackSchedule.mTimeQueue.Clear();
 
   mPlaybackSchedule.Init(
      t0, t1, options, mCaptureSequences.empty() ? nullptr : &mRecordingSchedule );
 
   unsigned int playbackChannels = 0;
   size_t numCaptureChannels = 0;
   sampleFormat captureFormat = floatSample;
   double captureRate = 44100.0;
 
   auto pListener = GetListener();
 
   if (sequences.playbackSequences.size() > 0
      || sequences.otherPlayableSequences.size() > 0)
      playbackChannels = 2;
 
   if (mSoftwarePlaythrough)
      playbackChannels = 2;
 
   if (mCaptureSequences.size() > 0) {
      numCaptureChannels = accumulate(
         mCaptureSequences.begin(), mCaptureSequences.end(), size_t{},
         [](auto acc, const auto &pSequence) {
            return acc + pSequence->NChannels();
         });
      // I don't deal with the possibility of the capture sequences
      // 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 sequence was 16-bit and the second sequence was 24-bit,
      // we would set the sound card to capture in 16 bits and the second
      // sequence wouldn't get the benefit of all 24 bits the card is capable
      // of.
      const auto &sequence0 = mCaptureSequences[0];
      captureFormat = sequence0->GetSampleFormat();
      captureRate = sequence0->GetRate();
 
      // Tell project that we are about to start recording
      if (pListener)
         pListener->OnAudioIOStartRecording();
   }
 
   bool successAudio;
 
   mCaptureFormat = captureFormat;
   mCaptureRate = captureRate;
   successAudio =
      StartPortAudioStream(options, playbackChannels, numCaptureChannels);
 
   // Call this only after reassignment of mRate that might happen in the
   // previous call.
   mPlaybackSchedule.GetPolicy().Initialize( mPlaybackSchedule, mRate );
 
   auto range = Extensions();
   successAudio = successAudio &&
      std::all_of(range.begin(), range.end(),
         [this, &sequences, t0](auto &ext){
            return ext.StartOtherStream(sequences,
              (mPortStreamV19 != NULL && mLastPaError == paNoError)
                 ? Pa_GetStreamInfo(mPortStreamV19) : nullptr,
              t0, mRate ); });
 
   if (!successAudio) {
      if (pListener && numCaptureChannels > 0)
         pListener->OnAudioIOStopRecording();
      mStreamToken = 0;
 
      return 0;
   }
 
   {
      double mixerStart = t0;
      if (pStartTime)
         mixerStart = std::min( mixerStart, *pStartTime );
      if (!AllocateBuffers(options, sequences,
         mixerStart, mixerLimit, options.rate))
         return 0;
   }
 
   mpTransportState = std::make_unique<TransportState>(mOwningProject,
      mPlaybackSequences, mNumPlaybackChannels, 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.SetSequenceTime( time );
      mPlaybackSchedule.GetPolicy().OffsetSequenceTime( mPlaybackSchedule, 0 );
 
      // Reset mixer positions for all playback sequences
      for (auto &mixer : mPlaybackMixers)
         mixer->Reposition( time );
   }
 
   // Now that we are done with AllocateBuffers() and SetSequenceTime():
   mPlaybackSchedule.mTimeQueue.Prime(mPlaybackSchedule.GetSequenceTime());
   // else recording only without overdub
 
   // We signal the audio thread to call SequenceBufferExchange, to prime the RingBuffers
   // so that they will have data in them when the stream starts.  Having the
   // audio thread call SequenceBufferExchange here makes the code more predictable, since
   // SequenceBufferExchange will ALWAYS get called from the Audio thread.
   mAudioThreadShouldCallSequenceBufferExchangeOnce
      .store(true, std::memory_order_release);
 
   while( mAudioThreadShouldCallSequenceBufferExchangeOnce
      .load(std::memory_order_acquire)) {
      using namespace std::chrono;
      auto interval = 50ms;
      if (options.playbackStreamPrimer) {
         interval = options.playbackStreamPrimer();
      }
      std::this_thread::sleep_for(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 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.
      StartAudioThread();
 
      mForceFadeOut.store(false, std::memory_order_relaxed);
 
      // Now start the PortAudio stream!
      PaError err;
      err = Pa_StartStream( mPortStreamV19 );
 
      if( err != paNoError )
      {
         mStreamToken = 0;
 
         StopAudioThread();
 
         if (pListener && mNumCaptureChannels > 0)
            pListener->OnAudioIOStopRecording();
         StartStreamCleanup();
         // PRL: PortAudio error messages are sadly not internationalized
         BasicUI::ShowMessageBox(
            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)
      Publish({ pOwningProject.get(), AudioIOEvent::PLAYBACK, true });
   if (mNumCaptureChannels > 0)
      Publish({ pOwningProject.get(), AudioIOEvent::CAPTURE, true });
 
   commit = true;
 
   WaitForAudioThreadStarted();
 
   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.)
   BasicUI::CallAfter(move(action));
}
 
bool AudioIO::AllocateBuffers(
   const AudioIOStartStreamOptions &options,
   const TransportSequences &sequences, 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
   // sequences.  This allows us to scale up to many sequences 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.count() * 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), mNumCaptureChannels);
   mMinCaptureSecsToCopy =
      0.2 + 0.2 * std::min(size_t(16), mNumCaptureChannels);
 
   bool bDone;
   do
   {
      bDone = true; // assume success
      try
      {
         if( mNumPlaybackChannels > 0 ) {
            // Allocate output buffers.
            // Allow at least 2x of the buffer latency.
            auto playbackBufferSize =
               std::max((size_t)lrint(mRate * mPlaybackRingBufferSecs.count()), mHardwarePlaybackLatencyFrames * 2);
 
            // Make playbackBufferSize a multiple of mPlaybackSamplesToCopy
            playbackBufferSize = mPlaybackSamplesToCopy *
               ((playbackBufferSize + mPlaybackSamplesToCopy - 1) / mPlaybackSamplesToCopy);
 
            // Adjust mPlaybackRingBufferSecs correspondingly
            mPlaybackRingBufferSecs = PlaybackPolicy::Duration { playbackBufferSize / mRate };
 
            mPlaybackBuffers.resize(0);
            mProcessingBuffers.resize(0);
            mMasterBuffers.resize(0);
 
            // Always make at least one playback buffer, in case of
            // MIDI playback without any audio
            if(mPlaybackSequences.empty())
               mPlaybackBuffers.resize(1);
            else
            {
               mPlaybackBuffers.resize(mNumPlaybackChannels);
               mProcessingBuffers.resize(std::accumulate(
                  mPlaybackSequences.begin(),
                  mPlaybackSequences.end(),
                  0, [](int n, auto& seq) { return n + seq->NChannels(); }
               ));
               for(auto& buffer : mProcessingBuffers)
                  buffer.reserve(playbackBufferSize);
 
               mMasterBuffers.resize(mNumPlaybackChannels);
               for(auto& buffer : mMasterBuffers)
                  buffer.reserve(playbackBufferSize);
 
               // Number of scratch buffers depends on device playback channels
               if (mNumPlaybackChannels > 0) {
                  mScratchBuffers.resize(mNumPlaybackChannels * 2 + 1);
                  mScratchPointers.clear();
                  for (auto &buffer : mScratchBuffers) {
                     buffer.Allocate(playbackBufferSize, floatSample);
                     mScratchPointers.push_back(
                        reinterpret_cast<float*>(buffer.ptr()));
                  }
               }
            }
 
            std::generate(
               mPlaybackBuffers.begin(),
               mPlaybackBuffers.end(),
               [=]{ return std::make_unique<RingBuffer>(floatSample, playbackBufferSize); }
            );
 
            mPlaybackMixers.clear();
 
            const auto &warpOptions =
               policy.MixerWarpOptions(mPlaybackSchedule);
 
            mPlaybackQueueMinimum = lrint( mRate * times.latency.count() );
            mPlaybackQueueMinimum =
               std::min( mPlaybackQueueMinimum, playbackBufferSize );
 
            // Limit the mPlaybackQueueMinimum to the hardware latency
            mPlaybackQueueMinimum =
               std::max(mPlaybackQueueMinimum, mHardwarePlaybackLatencyFrames);
 
            // Make mPlaybackQueueMinimum a multiple of mPlaybackSamplesToCopy
            mPlaybackQueueMinimum = mPlaybackSamplesToCopy *
               ((mPlaybackQueueMinimum + mPlaybackSamplesToCopy - 1) / mPlaybackSamplesToCopy);
 
            // Bug 1763 - We must fade in from zero to avoid a click on starting.
            mOldPlaybackGain = 0.0f;
            for (unsigned int i = 0; i < mPlaybackSequences.size(); i++) {
               const auto &pSequence = mPlaybackSequences[i];
 
               // By the precondition of StartStream which is sole caller of
               // this function:
               assert(pSequence->FindChannelGroup());
               // use sequence time for the end time, not real time!
               double startTime, endTime;
               if (!sequences.prerollSequences.empty())
                  startTime = mPlaybackSchedule.mT0;
               else
                  startTime = t0;
 
               if (make_iterator_range(sequences.prerollSequences)
                  .contains(pSequence))
                  // Stop playing this sequence after pre-roll
                  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
                  endTime = t1;
 
               Mixer::Inputs mixSequences;
               mixSequences.push_back(Mixer::Input{ pSequence });
               mPlaybackMixers.emplace_back(std::make_unique<Mixer>(
                  std::move(mixSequences), std::nullopt,
                  // Don't throw for read errors, just play silence:
                  false, warpOptions, startTime, endTime,
                  pSequence->NChannels(),
                  std::max(mPlaybackSamplesToCopy, mPlaybackQueueMinimum),
                  false, // not interleaved
                  mRate, floatSample,
                  false,   // low quality dithering and resampling
                  nullptr, // no custom mix-down
                  Mixer::ApplyGain::Discard // don't apply gains
                  ));
            }
 
            const auto timeQueueSize = 1 +
               (playbackBufferSize + TimeQueueGrainSize - 1)
                  / TimeQueueGrainSize;
            mPlaybackSchedule.mTimeQueue.Init( timeQueueSize );
         }
 
         if( mNumCaptureChannels > 0 )
         {
            // Allocate input buffers.  For every input sequence 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)
            {
               BasicUI::ShowMessageBox( XO("Out of memory!") );
               return false;
            }
 
            mCaptureBuffers.resize(0);
            mCaptureBuffers.resize(mNumCaptureChannels);
            mResample.resize(0);
            mResample.resize(mNumCaptureChannels);
            mFactor = sampleRate / mRate;
 
            for (unsigned int i = 0; i < mNumCaptureChannels; ++i) {
               mCaptureBuffers[i] = std::make_unique<RingBuffer>(
                  mCaptureFormat, 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.count());
         if(playbackBufferSize < 100 || mPlaybackSamplesToCopy < 100)
         {
            BasicUI::ShowMessageBox( XO("Out of memory!") );
            return false;
         }
      }
   } while(!bDone);
 
   success = true;
   return true;
}
 
void AudioIO::StartStreamCleanup(bool bOnlyBuffers)
{
   mpTransportState.reset();
 
   mPlaybackBuffers.clear();
   mScratchBuffers.clear();
   mScratchPointers.clear();
   mPlaybackMixers.clear();
   mCaptureBuffers.clear();
   mResample.clear();
   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);
}
 
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( mAudioThreadSequenceBufferExchangeLoopRunning
      .load(std::memory_order_relaxed) )
   {
      // 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 )) {
         using namespace std::chrono;
         std::this_thread::sleep_for(milliseconds{latency + 50});
      }
   }
 
   wxMutexLocker locker(mSuspendAudioThread);
 
   //
   // 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+
 
   StopAudioThread();
 
   // 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;
   }
 
 
 
   // We previously told AudioThread to stop processing, now let's
   // be sure it has really stopped before resetting mpTransportState
   WaitForAudioThreadStopped();
 
 
   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 RecordableSequence.  To do this, we ask the audio thread
      // to call SequenceBufferExchange one last time (it normally would not do
      // so since Pa_GetStreamActive() would now return false
      ProcessOnceAndWait();
   }
 
   // No longer need effects processing. This must be done after the stream is stopped
   // to prevent the callback from being invoked after the effects are finalized.
   mpTransportState.reset();
 
   //
   // Everything is taken care of.  Now, just free all the resources
   // we allocated in StartStream()
   //
   mPlaybackBuffers.clear();
   mScratchBuffers.clear();
   mScratchPointers.clear();
   mPlaybackMixers.clear();
   mPlaybackSchedule.mTimeQueue.Clear();
 
   if (mStreamToken > 0)
   {
      //
      // Offset all recorded sequences to account for latency
      //
      if (mCaptureSequences.size() > 0) {
         mCaptureBuffers.clear();
         mResample.clear();
 
         //
         // We only apply latency correction when we actually played back
         // sequences during the recording. If we did not play back sequences,
         // 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 sequence in a project.
         //
 
         for (auto &sequence : mCaptureSequences) {
            // 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 sequence, at the top of the
            // guarded call, relying on the guarantee that the sequence will be
            // left in a flushed state, though the append buffer may be lost.
 
            GuardedCall( [&] {
               // use No-fail-guarantee that sequence is flushed,
               // Partial-guarantee that some initial length of the recording
               // is saved.
               // See comments in SequenceBufferExchange().
               sequence->Flush();
            } );
         }
 
 
         if (!mLostCaptureIntervals.empty()) {
            // This scope may combine many insertions of silence
            // into one transaction, lessening the number of checkpoints
            std::optional<TransactionScope> pScope;
            if (auto pOwningProject = mOwningProject.lock())
               pScope.emplace(*pOwningProject, "Dropouts");
            for (auto &interval : mLostCaptureIntervals) {
               auto &start = interval.first;
               auto duration = interval.second;
               for (auto &sequence : mCaptureSequences)
                  GuardedCall([&] {
                     sequence->InsertSilence(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();
 
   if (pListener && mNumCaptureChannels > 0)
      pListener->OnAudioIOStopRecording();
 
   BasicUI::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;
 
   {
      auto pOwningProject = mOwningProject.lock();
      if (mNumPlaybackChannels > 0)
         Publish({ pOwningProject.get(), AudioIOEvent::PLAYBACK, false });
      if (mNumCaptureChannels > 0)
         Publish({ pOwningProject.get(),
            wasMonitoring
               ? AudioIOEvent::MONITOR
               : AudioIOEvent::CAPTURE,
            false });
   }
 
   ResetOwningProject();
 
   mNumCaptureChannels = 0;
   mNumPlaybackChannels = 0;
 
   mPlaybackSequences.clear();
   mCaptureSequences.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, bool publish)
{
   if (state != IsPaused())
   {
      if (auto pOwningProject = mOwningProject.lock()) {
         // The realtime effects manager may remain "active" but becomes
         // "suspended" or "resumed".
         auto &em = RealtimeEffectManager::Get(*pOwningProject);
         em.SetSuspended(state);
      }
   }
 
   mPaused.store(state, std::memory_order_relaxed);
 
   if (publish)
      Publish({ mOwningProject.lock().get(), AudioIOEvent::PAUSE, 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()
{
   // Sequence time readout for the main thread
 
   if( !IsStreamActive() )
      return BAD_STREAM_TIME;
 
   return mPlaybackSchedule.GetSequenceTime();
}
 
 
//
//     Audio Thread Context
//
 
void AudioIO::AudioThread(std::atomic<bool> &finish)
{
   enum class State { eUndefined, eOnce, eLoopRunning, eDoNothing, eMonitoring } lastState = State::eUndefined;
   AudioIO *const gAudioIO = AudioIO::Get();
   while (!finish.load(std::memory_order_acquire)) {
      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->mAudioThreadSequenceBufferExchangeLoopActive
         .store(true, std::memory_order_relaxed);
      if( gAudioIO->mAudioThreadShouldCallSequenceBufferExchangeOnce
         .load(std::memory_order_acquire) )
      {
         gAudioIO->SequenceBufferExchange();
         gAudioIO->mAudioThreadShouldCallSequenceBufferExchangeOnce
            .store(false, std::memory_order_release);
 
         lastState = State::eOnce;
      }
      else if( gAudioIO->mAudioThreadSequenceBufferExchangeLoopRunning
         .load(std::memory_order_relaxed))
      {
         if (lastState != State::eLoopRunning)
         {
            // Main thread has told us to start - acknowledge that we do
            gAudioIO->mAudioThreadAcknowledge.store(Acknowledge::eStart,
                                                    std::memory_order::memory_order_release);
         }
         lastState = State::eLoopRunning;
 
         // We call the processing after raising the acknowledge flag, because the main thread
         // only needs to know that the message was seen.
         //
         // This is unlike the case with mAudioThreadShouldCallSequenceBufferExchangeOnce where the
         // store really means that the one-time exchange was done.
 
         gAudioIO->SequenceBufferExchange();
      }
      else
      {
         if (    (lastState == State::eLoopRunning)
              || (lastState == State::eMonitoring ) )
         {
            // Main thread has told us to stop; (actually: to neither process "once" nor "loop running")
            // acknowledge that we received the order and that no more processing will be done.
            gAudioIO->mAudioThreadAcknowledge.store(Acknowledge::eStop,
                                                    std::memory_order::memory_order_release);
         }
         lastState = State::eDoNothing;
 
         if (gAudioIO->IsMonitoring())
         {
            lastState = State::eMonitoring;
         }
      }
 
      gAudioIO->mAudioThreadSequenceBufferExchangeLoopActive
         .store(false, std::memory_order_relaxed);
 
      std::this_thread::sleep_until( loopPassStart + interval );
   }
}
 
size_t AudioIoCallback::MinValue(
   const RingBuffers &buffers, size_t (RingBuffer::*pmf)() const)
{
   return std::accumulate(buffers.begin(), buffers.end(),
      std::numeric_limits<size_t>::max(),
      [pmf](auto value, auto &pBuffer){
         return std::min(value, (pBuffer.get()->*pmf)()); });
}
 
size_t AudioIO::GetCommonlyFreePlayback()
{
   auto commonlyAvail = MinValue(mPlaybackBuffers, &RingBuffer::AvailForPut);
   // MB: subtract a few samples because the code in SequenceBufferExchange has rounding
   // errors
   return commonlyAvail - std::min(size_t(10), commonlyAvail);
}
 
size_t AudioIoCallback::GetCommonlyReadyPlayback()
{
   return MinValue(mPlaybackBuffers, &RingBuffer::AvailForGet);
}
 
size_t AudioIoCallback::GetCommonlyWrittenForPlayback()
{
   return MinValue(mPlaybackBuffers, &RingBuffer::WrittenForGet);
}
 
size_t AudioIO::GetCommonlyAvailCapture()
{
   return MinValue(mCaptureBuffers, &RingBuffer::AvailForGet);
}
 
// 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::SequenceBufferExchange()
{
   FillPlayBuffers();
   DrainRecordBuffers();
}
 
void AudioIO::FillPlayBuffers()
{
   std::optional<RealtimeEffects::ProcessingScope> pScope;
   if (mpTransportState && mpTransportState->mpRealtimeInitialization)
      pScope.emplace(
         *mpTransportState->mpRealtimeInitialization, mOwningProject);
 
   if (mNumPlaybackChannels == 0)
      return;
 
   // 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.  Also, if in a previous pass, unequal numbers of samples were
   // discarded from ring buffers for differing latencies.
 
   // To keep things simple, we write no more data than 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
   // at least mPlaybackSamplesToCopy.  This improves performance
   // by not always trying to process tiny chunks, eating the
   // CPU unnecessarily.
   if (nAvailable < mPlaybackSamplesToCopy)
      return;
 
   // 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 GetNeeded = [&]() -> size_t {
      // Note that reader might concurrently consume between loop passes below
      // So this might not be nondecreasing
      auto nReady = GetCommonlyWrittenForPlayback();
      return mPlaybackQueueMinimum - std::min(mPlaybackQueueMinimum, nReady);
   };
   auto nNeeded = GetNeeded();
 
   // wxASSERT( nNeeded <= nAvailable );
 
   auto Flush = [&]{
      /* The flushing of all the Puts to the RingBuffers is lifted out of the
      do-loop in ProcessPlaybackSlices, and also after transformation of the
      stream for realtime effects.
 
      It's only here that a release is done on the atomic variable that
      indicates the readiness of sample data to the consumer.  That atomic
      also synchronizes the use of the TimeQueue.
      */
      for (const auto &pBuffer : mPlaybackBuffers)
         pBuffer->Flush();
   };
 
   while (true) {
      // Limit maximum buffer size (increases performance)
      auto available = std::min( nAvailable,
         std::max( nNeeded, mPlaybackSamplesToCopy ) );
 
      // After each loop pass or after break
      Finally Do{ Flush };
 
      if (!ProcessPlaybackSlices(pScope, available))
         // We are not making progress.  May fail to satisfy the minimum but
         // won't loop forever
         break;
 
      // Loop again to satisfy the minimum queue requirement in case there
      // was discarding of processed data for effect latencies
      nNeeded = GetNeeded();
      if (nNeeded == 0)
         break;
 
      // Might increase because the reader consumed some
      nAvailable = GetCommonlyFreePlayback();
   }
}
 
#define stackAllocate(T, count) static_cast<T*>(alloca(count * sizeof(T)))
 
bool AudioIO::ProcessPlaybackSlices(
   std::optional<RealtimeEffects::ProcessingScope> &pScope, size_t available)
{
   auto &policy = mPlaybackSchedule.GetPolicy();
 
   // 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;
   bool progress = false;
 
   // remember initial processing buffer offsets
   // they may be different depending on latencies
   const auto processingBufferOffsets = stackAllocate(size_t, mProcessingBuffers.size());
   for(unsigned n = 0; n < mProcessingBuffers.size(); ++n)
      processingBufferOffsets[n] = mProcessingBuffers[n].size();
 
   do {
      const auto slice =
         policy.GetPlaybackSlice(mPlaybackSchedule, available);
      const auto &[frames, toProduce] = slice;
      progress = progress || toProduce > 0;
 
      // 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, slice);
 
      // mPlaybackMixers correspond one-to-one with mPlaybackSequences
      size_t iSequence = 0;
      // mPlaybackBuffers correspond many-to-one with mPlaybackSequences
      size_t iBuffer = 0;
      for (auto &mixer : mPlaybackMixers) {
         // 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 = mixer->Process(toProduce);
 
            //wxASSERT(produced <= toProduce);
            // Copy (non-interleaved) mixer outputs to one or more ring buffers
            const auto nChannels = mPlaybackSequences[iSequence]->NChannels();
 
            const auto appendPos = mProcessingBuffers[iBuffer].size();
            for (size_t j = 0; j < nChannels; ++j)
            {
               // mPlaybackBuffers correspond many-to-one with mPlaybackSequences
               auto& buffer = mProcessingBuffers[iBuffer + j];
               //Sufficient size should have been reserved in AllocateBuffers
               //But for some latency values (> aprox. 100ms) pre-allocated
               //buffer could be not large enough.
               //Preserve what was written to the buffer during previous pass, don't discard
               buffer.resize(buffer.size() + frames, 0);
 
               const auto warpedSamples = mixer->GetBuffer(j);
               std::copy_n(
                  reinterpret_cast<const float*>(warpedSamples),
                  produced,
                  buffer.data() + appendPos);
               std::fill_n(
                  buffer.data() + appendPos + produced,
                  frames - produced,
                  .0f);
            }
 
            iBuffer += nChannels;
            ++iSequence;
         }
      }
 
      available -= frames;
      // wxASSERT(available >= 0); // don't assert on this thread
      if(mPlaybackSequences.empty())
         // Produce silence in the single ring buffer
         mPlaybackBuffers[0]->Put(nullptr, floatSample, 0, frames);
 
      done = policy.RepositionPlayback( mPlaybackSchedule, mPlaybackMixers,
         frames, available );
   } while (available && !done);
 
   //stop here if there are no sample sources to process...
   if(mPlaybackSequences.empty())
      return progress;
 
   // Do any realtime effect processing for each individual sample source,
   // after all the little slices have been written.
   if (pScope)
   {
      const auto pointers = stackAllocate(float*, mNumPlaybackChannels);
 
      int bufferIndex = 0;
      for(const auto& seq : mPlaybackSequences)
      {
         if(!seq)
            continue;//no similar check in convert-to-float part
         const auto channelGroup = seq->FindChannelGroup();
         if(!channelGroup)
            continue;
 
         // Are there more output device channels than channels of vt?
         // Such as when a mono sequence is processed for stereo play?
         // Then supply some non-null fake input buffers, because the
         // various ProcessBlock overrides of effects may crash without it.
         // But it would be good to find the fixes to make this unnecessary.
         auto scratch = &mScratchPointers[mNumPlaybackChannels + 1];
 
         //skip samples that are already processed
         const auto offset = processingBufferOffsets[bufferIndex];
         //number of newly written samples
         const auto len = mProcessingBuffers[bufferIndex].size() - offset;
 
         if(len > 0)
         {
            for(unsigned i = 0; i < seq->NChannels(); ++i)
               pointers[i] = mProcessingBuffers[bufferIndex + i].data() + offset;
 
            for(unsigned i = seq->NChannels(); i < mNumPlaybackChannels; ++i)
            {
 
               pointers[i] = *scratch++;
               std::fill_n(pointers[i], len, .0f);
            }
 
            const auto discardable = pScope->Process(channelGroup, &pointers[0],
               mScratchPointers.data(),
               // The single dummy output buffer:
               mScratchPointers[mNumPlaybackChannels],
               mNumPlaybackChannels, len);
            // Check for asynchronous user changes in mute, solo status
            const auto silenced = SequenceShouldBeSilent(*seq);
            for(int i = 0; i < seq->NChannels(); ++i)
            {
               auto& buffer = mProcessingBuffers[bufferIndex + i];
               buffer.erase(buffer.begin() + offset, buffer.begin() + offset + discardable);
               if(silenced)
               {
                  //TODO: fade out smoothly
                  std::fill_n(buffer.data() + offset, len - discardable, 0);
               }
            }
         }
 
         bufferIndex += seq->NChannels();
      }
   }
 
   //samples at the beginning could have been discarded
   //in the previous step, proceed with the number of samples
   //equal to the shortest buffer size available
   auto samplesAvailable = std::min_element(
      mProcessingBuffers.begin(),
      mProcessingBuffers.end(),
      [](auto& first, auto& second) { return first.size() < second.size(); }
   )->size();
 
   //introduced latency may be too high...
   //but we don't discard what already was written
   if(samplesAvailable == 0)
      return progress;
 
   //Prepare master buffers.
   auto cleanup = finally([=] {
      for(auto& buffer : mMasterBuffers)
         buffer.clear();
   });
 
   for(auto& buffer : mMasterBuffers)
   {
      //assert(buffer.size() == 0);
      //assert(buffer.capacity() >= samplesAvailable);
      buffer.resize(samplesAvailable, 0);
   }
 
   {
      unsigned bufferIndex = 0;
      for(const auto& seq : mPlaybackSequences)
      {
         //TODO: apply micro-fades
         const auto numChannels = seq->NChannels();
         if(numChannels > 1)
         {
            for(unsigned n = 0; n < seq->NChannels(); ++n)
            {
               const auto gain = seq->GetChannelGain(n);
               for(unsigned i = 0; i < samplesAvailable; ++i)
                  mMasterBuffers[n][i] += mProcessingBuffers[bufferIndex + n][i] * gain;
            }
         }
         else if(numChannels == 1)
         {
            //mono source is duplicated into every output channel
            for(unsigned n = 0; n < mNumPlaybackChannels; ++n)
            {
               const auto gain = seq->GetChannelGain(n);
               for(unsigned i = 0; i < samplesAvailable; ++i)
                  mMasterBuffers[n][i] += mProcessingBuffers[bufferIndex][i] * gain;
            }
         }
         bufferIndex += seq->NChannels();
      }
   }
 
   //remove only samples that were processed in previous step
   for(auto& buffer : mProcessingBuffers)
      buffer.erase(buffer.begin(), buffer.begin() + samplesAvailable);
 
   // Do any realtime effect processing, after all the little
   // slices have been written. This time we use mixed source created in
   // previous step
   size_t masterBufferOffset = 0;//The amount of samples to be discarded
   if(pScope)
   {
      const auto pointers = stackAllocate(float*, mNumPlaybackChannels);
      for(unsigned i = 0; i < mNumPlaybackChannels; ++i)
         pointers[i] = mMasterBuffers[i].data();
 
      masterBufferOffset = pScope->Process(
         RealtimeEffectManager::MasterGroup,
         &pointers[0],
         mScratchPointers.data(),
         // The single dummy output buffer:
         mScratchPointers[mNumPlaybackChannels],
         mNumPlaybackChannels, samplesAvailable);
 
      // wxASSERT(samplesAvailable >= masterBufferOffset); // don't assert on this thread
      samplesAvailable -= masterBufferOffset;
   }
 
   if(samplesAvailable == 0)
      return progress;
 
   {
      unsigned bufferIndex = 0;
      for(auto& buffer : mMasterBuffers)
      {
         mPlaybackBuffers[bufferIndex++]->Put(
            reinterpret_cast<constSamplePtr>(buffer.data()) + masterBufferOffset * sizeof(float),
            floatSample,
            samplesAvailable,
            0
         );
      }
   }
 
   for(auto& buffer : mMasterBuffers)
      buffer.clear();
 
   return progress;
}
 
void AudioIO::DrainRecordBuffers()
{
   if (mRecordingException || mCaptureSequences.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 RecordableSequence 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 RecordableSequences, after the failed append operation, that
      // the sequences 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 );
      for (auto &pSequence: mCaptureSequences)
         pSequence->RepairChannels();
   };
 
   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 (mAudioThreadShouldCallSequenceBufferExchangeOnce
          .load(std::memory_order_relaxed) ||
          deltat >= mMinCaptureSecsToCopy)
      {
         bool newBlocks = false;
 
         // Append captured samples to the end of the RecordableSequences.
         // (WaveTracks have their own buffering for efficiency.)
         auto iter = mCaptureSequences.begin();
         auto width = (*iter)->NChannels();
         size_t iChannel = 0;
         for (size_t i = 0; i < mNumCaptureChannels; ++i) {
            Finally Do {[&]{
               if (++iChannel == width) {
                  ++iter;
                  iChannel = 0;
                  if (iter != mCaptureSequences.end())
                     width = (*iter)->NChannels();
               }
            }};
            size_t discarded = 0;
 
            if (!mRecordingSchedule.mLatencyCorrected) {
               const auto correction = mRecordingSchedule.TotalCorrection();
               if (correction >= 0) {
                  // Rightward shift
                  // Once only (per sequence per recording), insert some initial
                  // silence.
                  size_t size = floor( correction * mRate * mFactor);
                  SampleBuffer temp(size, mCaptureFormat);
                  ClearSamples(temp.ptr(), mCaptureFormat, 0, size);
                  (*iter)->Append(iChannel, temp.ptr(), mCaptureFormat, size, 1,
                     // Do not dither recordings
                     narrowestSampleFormat);
               }
               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
               // sequence
               const auto &data = mRecordingSchedule.mCrossfadeData[i];
               totalCrossfadeLength = data.size();
               if (totalCrossfadeLength) {
                  crossfadeStart =
                     floor(mRecordingSchedule.Consumed() * mCaptureRate);
                  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 = mCaptureFormat;
               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 = (*iter)->Append(iChannel,
               temp.ptr(), format, size, 1,
               // Do not dither recordings
               narrowestSampleFormat
            ) || 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();
 
      }
      // 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.GetSequenceTime() - 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.GetSequenceTime();
 
         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
 
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();
   }
}
 
// Limit values to -1.0..+1.0
void ClampBuffer(float * pBuffer, unsigned long len){
   for(unsigned i = 0; i < len; i++)
      pBuffer[i] = std::clamp(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 *outputFloats,
   unsigned long framesPerBuffer,
   float *outputMeterFloats
)
{
   const auto numPlaybackSequences = mPlaybackSequences.size();
   const auto numPlaybackChannels = mNumPlaybackChannels;
 
   mMaxFramesOutput = 0;
 
   // Quick returns if next to nothing to do.
   if (mStreamToken <= 0 ||
       !outputFloats ||
       numPlaybackChannels <= 0) {
      // So that UpdateTimePosition() will be correct, in case of MIDI play with
      // no audio output channels
      mMaxFramesOutput = framesPerBuffer;
      return false;
   }
 
   if (mSeek && !mPlaybackSchedule.GetPolicy().AllowSeek(mPlaybackSchedule))
      mSeek = 0.0;
 
   if (mSeek){
      mCallbackReturn = CallbackDoSeek();
      return true;
   }
 
   // Choose a common size to take from all ring buffers
   const auto toGet =
      std::min<size_t>(framesPerBuffer, GetCommonlyReadyPlayback());
 
   // Poke: If there are no playback sequences, then check playback
   // completion condition and do early return
   // PRL:  Also consume frbom the single playback ring buffer
   if (numPlaybackSequences == 0) {
      mMaxFramesOutput = mPlaybackBuffers[0]->Discard(toGet);
      CallbackCheckCompletion(mCallbackReturn, 0);
      mLastPlaybackTimeMillis = ::wxGetUTCTimeMillis();
      return false;
   }
 
   // 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.
 
   // ------ MEMORY ALLOCATION ----------------------
   // These are small structures.
   const auto tempBufs = stackAllocate(float *, numPlaybackChannels);
 
   // And these are larger structures....
   for (unsigned int c = 0; c < numPlaybackChannels; c++)
      tempBufs[c] = stackAllocate(float, framesPerBuffer);
   // ------ End of MEMORY ALLOCATION ---------------
 
   auto gain = ExpGain(GetMixerOutputVol());
   if (mForceFadeOut.load(std::memory_order_relaxed) || IsPaused())
      gain = 0.0;
 
   for(unsigned n = 0; n < numPlaybackChannels; ++n)
   {
      decltype(framesPerBuffer) len = mPlaybackBuffers[n]->Get(
         reinterpret_cast<samplePtr>(tempBufs[n]),
         floatSample,
         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 SequenceBufferExchange 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[n][len], 0,
            (framesPerBuffer - len) * sizeof(float));
      }
 
      // PRL:  More recent rewrites of SequenceBufferExchange 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);
 
      len = mMaxFramesOutput;
 
      // Realtime effect transformation of the sound used to happen here
      // but it is now done already on the producer side of the RingBuffer
 
      // Mix the results with the existing output (software playthrough) and
      // apply panning.  If post panning effects are desired, the panning would
      // need to be be split out from the mixing and applied in a separate step.
 
      // Our channels aren't silent.  We need to pass their data on.
      //
      // Each channel in the sequences 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)
      {
         // 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+n] +=
                  gain*tempBufs[n][i];
         }
 
         auto oldGain = mOldPlaybackGain;
         // if no microfades, jump in volume.
         if (!mbMicroFades)
            oldGain = gain;
 
         // Linear interpolate.
         // PRL todo:  choose denominator differently, so it doesn't depend on
         // framesPerBuffer, which is influenced by the portAudio implementation in
         // opaque ways
         const float deltaGain = (gain - oldGain) / len;
         for (unsigned i = 0; i < len; i++)
         {
            outputFloats[numPlaybackChannels * i + n] +=
               (oldGain + deltaGain * i) * tempBufs[n][i];
         }
      }
      CallbackCheckCompletion(mCallbackReturn, len);
   }
 
   mOldPlaybackGain = gain;
 
   // wxASSERT( maxLen == toGet );
 
   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.SetSequenceTime(
      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 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 sequences, 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 SequenceBufferExchange, 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.load(std::memory_order_relaxed)
           && 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.GetSequenceTime() +
            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 = std::clamp(tmp, -32768.0f, 32767.0f);
               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);
      mCaptureBuffers[t]->Flush();
   }
}
 
 
#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;
   pInputMeter->UpdateDisplay(
      numCaptureChannels, framesPerBuffer, inputSamples);
}
 
/* 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;
   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 track panel timer events
      //MixerBoard* pMixerBoard = mOwningProject->GetMixerBoard();
      //if (pMixerBoard)
      //   pMixerBoard->UpdateMeters(GetStreamTime(),
      //                              (pProj->GetControlToolBar()->GetLastPlayMode() == loopedPlay));
}
 
unsigned AudioIoCallback::CountSoloingSequences(){
   const auto numPlaybackSequences = mPlaybackSequences.size();
 
   // MOVE_TO: CountSoloingSequences() function
   unsigned numSolo = 0;
   for (unsigned t = 0; t < numPlaybackSequences; t++ )
      if (mPlaybackSequences[t]->GetSolo())
         numSolo++;
   auto range = Extensions();
   numSolo += std::accumulate(range.begin(), range.end(), 0,
      [](unsigned sum, auto &ext){
         return sum + ext.CountOtherSolo(); });
   return numSolo;
}
 
// TODO: Consider making the two sequence status functions into member
// functions of sequence objects
 
// true IFF the sequence should be silent.
// The sequence may not yet be silent, since it may still be
// fading out.
bool AudioIoCallback::SequenceShouldBeSilent(const PlayableSequence &ps)
{
   return !ps.GetSolo() && (
      // Cut if somebody else is soloing
      mbHasSoloSequences ||
      // Cut if we're muted (and not soloing)
      ps.GetMute()
   );
}
 
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 sequences for change of state.
   // (User might click mute and solo buttons.)
   mbHasSoloSequences = CountSoloingSequences() > 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, IsPaused(),
         timeInfo,
         framesPerBuffer);
      ext.FillOtherBuffers(
         mRate, mNumPauseFrames, IsPaused(), mbHasSoloSequences);
   }
 
   // ------ 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;
   const auto tempFloats = stackAllocate(float,
      framesPerBuffer * std::max(numCaptureChannels, numPlaybackChannels));
 
   bool bVolEmulationActive =
      (outputBuffer && GetMixerOutputVol() != 1.0);
   // outputMeterFloats is the scratch pad for the output meter.
   // we can often reuse the existing outputBuffer and save on allocating
   // something new.
   const auto outputMeterFloats = bVolEmulationActive
      ? stackAllocate(float, framesPerBuffer * numPlaybackChannels)
      : 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 sequence audio to play (because we are paused and have faded
   // out)
   if( IsPaused() &&  (( !mbMicroFades ) || mOldPlaybackGain == 0.0f ))
      return mCallbackReturn;
 
   // To add sequence 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 sequence (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;
 
   // Pause audio thread and wait for it to finish
   //
   // [PM] the following 8 lines of code could be probably replaced by
   // a single call to StopAudioThreadAndWait()
   //
   // CAUTION: when trying the above, you must also replace the setting of the
   // atomic before the return, with a call to StartAudioThread()
   //
   // If that works, then we can remove mAudioThreadSequenceBufferExchangeLoopActive,
   // as it will become unused; consequently, the AudioThread loop would get simpler too.
   //
   mAudioThreadSequenceBufferExchangeLoopRunning
      .store(false, std::memory_order_relaxed);
 
   while( mAudioThreadSequenceBufferExchangeLoopActive
      .load(std::memory_order_relaxed ) )
   {
      using namespace std::chrono;
      std::this_thread::sleep_for(50ms);
   }
 
   // Calculate the NEW time position, in the PortAudio callback
   const auto time =
      mPlaybackSchedule.GetPolicy().OffsetSequenceTime( mPlaybackSchedule, mSeek );
 
   mPlaybackSchedule.SetSequenceTime( time );
   mSeek = 0.0;
 
 
   // Reset mixer positions and flush buffers for all sequences
   for (auto &mixer : mPlaybackMixers)
      mixer->Reposition( time, true );
   for (auto &buffer : mPlaybackBuffers) {
      const auto toDiscard = buffer->AvailForGet();
      const auto discarded = buffer->Discard( toDiscard );
      // wxASSERT( discarded == toDiscard );
      // but we can't assert in this thread
      wxUnusedVar(discarded);
   }
 
   mPlaybackSchedule.mTimeQueue.Prime(time);
 
   // Reload the ring buffers
   ProcessOnceAndWait();
 
   // Reenable the audio thread
   mAudioThreadSequenceBufferExchangeLoopRunning
      .store(true, std::memory_order_relaxed);
 
   return paContinue;
}
 
void AudioIoCallback::CallbackCheckCompletion(
   int &callbackReturn, unsigned long len)
{
   if (IsPaused())
      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());
}
 
 
void AudioIoCallback::StartAudioThread()
{
   mAudioThreadSequenceBufferExchangeLoopRunning.store(true, std::memory_order_release);
}
 
void AudioIoCallback::WaitForAudioThreadStarted()
{
   while (mAudioThreadAcknowledge.load(std::memory_order_acquire) != Acknowledge::eStart)
   {
      using namespace std::chrono;
      std::this_thread::sleep_for(50ms);
   }
   mAudioThreadAcknowledge.store(Acknowledge::eNone, std::memory_order_release);
}
 
void AudioIoCallback::StopAudioThread()
{
   mAudioThreadSequenceBufferExchangeLoopRunning.store(false, std::memory_order_release);
}
 
void AudioIoCallback::WaitForAudioThreadStopped()
{
   while (mAudioThreadAcknowledge.load(std::memory_order_acquire) != Acknowledge::eStop)
   {
      using namespace std::chrono;
      std::this_thread::sleep_for(50ms);
   }
   mAudioThreadAcknowledge.store(Acknowledge::eNone, std::memory_order_release);
}
 
void AudioIoCallback::ProcessOnceAndWait(std::chrono::milliseconds sleepTime)
{
   mAudioThreadShouldCallSequenceBufferExchangeOnce
      .store(true, std::memory_order_release);
 
   while (mAudioThreadShouldCallSequenceBufferExchangeOnce
      .load(std::memory_order_acquire))
   {
      using namespace std::chrono;
      std::this_thread::sleep_for(sleepTime);
   }
}
 
 
 
bool AudioIO::IsCapturing() const
{
   // Includes a test of mTime, used in the main thread
   return IsStreamActive() &&
      GetNumCaptureChannels() > 0 &&
      mPlaybackSchedule.GetSequenceTime() >=
         mPlaybackSchedule.mT0 + mRecordingSchedule.mPreRoll;
}
 
BoolSetting SoundActivatedRecord{ "/AudioIO/SoundActivatedRecord", false };