MIDIPlay.cpp

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/**********************************************************************
 
Audacity: A Digital Audio Editor
 
@file MIDIPlay.cpp
@brief Inject added MIDI playback capability into Audacity's audio engine
 
Paul Licameli split from AudioIO.cpp and AudioIOBase.cpp
 
*//*****************************************************************/
#include "MIDIPlay.h"
#include "AudioIO.h"
 
#include "BasicUI.h"
#include "Prefs.h"
#include "portaudio.h"
#include <portmidi.h>
#include <porttime.h>
#include <thread>
 
#define ROUND(x) (int) ((x)+0.5)
 
class NoteTrack;
using NoteTrackConstArray = std::vector < std::shared_ptr< const NoteTrack > >;
 
namespace {
 
/*
 Adapt and rename the implementation of PaUtil_GetTime from commit
 c5d2c51bd6fe354d0ee1119ba932bfebd3ebfacc of portaudio
 */
#if defined( __APPLE__ )
 
#include <mach/mach_time.h>
 
/* Scaler to convert the result of mach_absolute_time to seconds */
static double machSecondsConversionScaler_ = 0.0;
 
/* Initialize it */
static struct InitializeTime { InitializeTime() {
   mach_timebase_info_data_t info;
   kern_return_t err = mach_timebase_info( &info );
   if( err == 0  )
       machSecondsConversionScaler_ = 1e-9 * (double) info.numer / (double) info.denom;
} } initializeTime;
 
static PaTime util_GetTime( void )
{
   return mach_absolute_time() * machSecondsConversionScaler_;
}
 
#elif defined( __WXMSW__ )
 
#include "profileapi.h"
#include "sysinfoapi.h"
#include "timeapi.h"
 
static int usePerformanceCounter_;
static double secondsPerTick_;
 
static struct InitializeTime { InitializeTime() {
    LARGE_INTEGER ticksPerSecond;
 
    if( QueryPerformanceFrequency( &ticksPerSecond ) != 0 )
    {
        usePerformanceCounter_ = 1;
        secondsPerTick_ = 1.0 / (double)ticksPerSecond.QuadPart;
    }
    else
    {
        usePerformanceCounter_ = 0;
    }
} } initializeTime;
 
static double util_GetTime( void )
{
    LARGE_INTEGER time;
 
    if( usePerformanceCounter_ )
    {
        /*
            Note: QueryPerformanceCounter has a known issue where it can skip forward
            by a few seconds (!) due to a hardware bug on some PCI-ISA bridge hardware.
            This is documented here:
            http://support.microsoft.com/default.aspx?scid=KB;EN-US;Q274323&
 
            The work-arounds are not very paletable and involve querying GetTickCount
            at every time step.
 
            Using rdtsc is not a good option on multi-core systems.
 
            For now we just use QueryPerformanceCounter(). It's good, most of the time.
        */
        QueryPerformanceCounter( &time );
        return time.QuadPart * secondsPerTick_;
    }
    else
    {
   #if defined(WINAPI_FAMILY) && (WINAPI_FAMILY == WINAPI_FAMILY_APP)
        return GetTickCount64() * .001;
   #else
        return timeGetTime() * .001;
   #endif
    }
}
 
#elif defined(HAVE_CLOCK_GETTIME)
 
#include <time.h>
 
static PaTime util_GetTime( void )
{
    struct timespec tp;
    clock_gettime(CLOCK_REALTIME, &tp);
    return (PaTime)(tp.tv_sec + tp.tv_nsec * 1e-9);
}
 
#else
 
#include <sys/time.h>
 
static PaTime util_GetTime( void )
{
    struct timeval tv;
    gettimeofday( &tv, NULL );
    return (PaTime) tv.tv_usec * 1e-6 + tv.tv_sec;
}
 
#endif
 
enum {
   // This is the least positive latency we can
   // specify to Pm_OpenOutput, 1 ms, which prevents immediate
   // scheduling of events:
   MIDI_MINIMAL_LATENCY_MS = 1
};
 
// return the system time as a double
static double streamStartTime = 0; // bias system time to small number
 
static double SystemTime(bool usingAlsa)
{
#ifdef __WXGTK__
   if (usingAlsa) {
      struct timespec now;
      // CLOCK_MONOTONIC_RAW is unaffected by NTP or adj-time
#ifdef FreeBSD
      clock_gettime(CLOCK_REALTIME, &now);
#else
      clock_gettime(CLOCK_MONOTONIC_RAW, &now);
#endif
      //return now.tv_sec + now.tv_nsec * 0.000000001;
      return (now.tv_sec + now.tv_nsec * 0.000000001) - streamStartTime;
   }
#else
   static_cast<void>(usingAlsa);//compiler food.
#endif
 
   return util_GetTime() - streamStartTime;
}
 
bool MIDIPlay::mMidiStreamActive = false;
bool MIDIPlay::mMidiOutputComplete = true;
 
AudioIOExt::RegisteredFactory sMIDIPlayFactory{
   [](const auto &playbackSchedule){
      return std::make_unique<MIDIPlay>(playbackSchedule);
   }
};
 
MIDIPlay::MIDIPlay(const PlaybackSchedule &schedule)
   : mPlaybackSchedule{ schedule }
{
#ifdef AUDIO_IO_GB_MIDI_WORKAROUND
   // Pre-allocate with a likely sufficient size, exceeding probable number of
   // channels
   mPendingNotesOff.reserve(64);
#endif
 
   PmError pmErr = Pm_Initialize();
 
   if (pmErr != pmNoError) {
      auto errStr =
              XO("There was an error initializing the midi i/o layer.\n");
      errStr += XO("You will not be able to play midi.\n\n");
      wxString pmErrStr = LAT1CTOWX(Pm_GetErrorText(pmErr));
      if (!pmErrStr.empty())
         errStr += XO("Error: %s").Format( pmErrStr );
      // 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 Midi"))
            .ButtonStyle(Button::Ok)
            .IconStyle(Icon::Error));
 
      // Same logic for PortMidi as described above for PortAudio
   }
}
 
MIDIPlay::~MIDIPlay()
{
   Pm_Terminate();
}
 
bool MIDIPlay::StartOtherStream(const TransportTracks &tracks,
   const PaStreamInfo* info, double, double rate)
{
   mMidiPlaybackTracks.clear();
   for (const auto &pTrack : tracks.otherPlayableTracks) {
      pTrack->TypeSwitch( [&](const NoteTrack *pNoteTrack){
         mMidiPlaybackTracks.push_back(
            pNoteTrack->SharedPointer<const NoteTrack>());
      } );
   }
 
   streamStartTime = 0;
   streamStartTime = SystemTime(mUsingAlsa);
 
   mNumFrames = 0;
   // we want this initial value to be way high. It should be
   // sufficient to assume AudioTime is zero and therefore
   // mSystemMinusAudioTime is SystemTime(), but we'll add 1000s
   // for good measure. On the first callback, this should be
   // reduced to SystemTime() - mT0, and note that mT0 is always
   // positive.
   mSystemMinusAudioTimePlusLatency =
      mSystemMinusAudioTime = SystemTime(mUsingAlsa) + 1000;
   mAudioOutLatency = 0.0; // set when stream is opened
   mCallbackCount = 0;
   mAudioFramesPerBuffer = 0;
 
   // We use audio latency to estimate how far ahead of DACS we are writing
   if (info) {
      // this is an initial guess, but for PA/Linux/ALSA it's wrong and will be
      // updated with a better value:
      mAudioOutLatency = info->outputLatency;
      mSystemMinusAudioTimePlusLatency += mAudioOutLatency;
   }
 
   // TODO: it may be that midi out will not work unless audio in or out is
   // active -- this would be a bug and may require a change in the
   // logic here.
 
   bool successMidi = true;
 
   if(!mMidiPlaybackTracks.empty()){
      successMidi = StartPortMidiStream(rate);
   }
 
   // On the other hand, if MIDI cannot be opened, we will not complain
   // return successMidi;
   return true;
}
 
void MIDIPlay::AbortOtherStream()
{
   mMidiPlaybackTracks.clear();
}
 
PmTimestamp MidiTime(void *pInfo)
{
   return static_cast<MIDIPlay*>(pInfo)->MidiTime();
}
 
// Set up state to iterate NoteTrack events in sequence.
// Sends MIDI control changes up to the starting point mT0
// if send is true. Output is delayed by offset to facilitate
// looping (each iteration is delayed more).
void MIDIPlay::PrepareMidiIterator(bool send, double startTime, double offset)
{
   mIterator.emplace(mPlaybackSchedule, *this,
      mMidiPlaybackTracks, startTime, offset, send);
}
 
Iterator::Iterator(
   const PlaybackSchedule &schedule, MIDIPlay &midiPlay,
   NoteTrackConstArray &midiPlaybackTracks,
   double startTime, double offset, bool send )
   : mPlaybackSchedule{ schedule }
   , mMIDIPlay{ midiPlay }
{
   // instead of initializing with an Alg_seq, we use begin_seq()
   // below to add ALL Alg_seq's.
   // Iterator not yet initialized, must add each track...
   for (auto &t : midiPlaybackTracks) {
      Alg_seq_ptr seq = &t->GetSeq();
      // mark sequence tracks as "in use" since we're handing this
      // off to another thread and want to make sure nothing happens
      // to the data until playback finishes. This is just a sanity check.
      seq->set_in_use(true);
      const void *cookie = t.get();
      it.begin_seq(seq,
         // casting away const, but allegro just uses the pointer opaquely
         const_cast<void*>(cookie), t->GetOffset() + offset);
   }
   Prime(send, startTime + offset);
}
 
Iterator::~Iterator()
{
   it.end();
}
 
void Iterator::Prime(bool send, double startTime)
{
   GetNextEvent(); // prime the pump for FillOtherBuffers
 
   // Start MIDI from current cursor position
   while (mNextEvent &&
          GetNextEventTime() < startTime) {
      if (send)
         /*
          hasSolo argument doesn't matter because midiStateOnly is true.
          "Fast-forward" all update events from the start of track to the given
          play start time so the notes sound with correct timbre whenever
          turned on.
          */
         OutputEvent(0, true, false);
      GetNextEvent();
   }
}
 
double Iterator::GetNextEventTime() const
{
   if (mNextEvent == &gAllNotesOff)
      return mNextEventTime - ALG_EPS;
   return mNextEventTime;
}
 
bool MIDIPlay::StartPortMidiStream(double rate)
{
#ifdef __WXGTK__
   // Duplicating a bit of AudioIO::StartStream
   // Detect whether ALSA is the chosen host, and do the various involved MIDI
   // timing compensations only then.
   mUsingAlsa = (AudioIOHost.Read() == L"ALSA");
#endif
 
   int i;
   int nTracks = mMidiPlaybackTracks.size();
   // Only start MIDI stream if there is an open track
   if (nTracks == 0)
      return false;
 
   //wxPrintf("StartPortMidiStream: mT0 %g mTime %g\n",
   //       mT0, mTime);
 
   /* get midi playback device */
   PmDeviceID playbackDevice = Pm_GetDefaultOutputDeviceID();
   auto playbackDeviceName = MIDIPlaybackDevice.Read();
   mSynthLatency = MIDISynthLatency_ms.Read();
   if (wxStrcmp(playbackDeviceName, wxT("")) != 0) {
      for (i = 0; i < Pm_CountDevices(); i++) {
         const PmDeviceInfo *info = Pm_GetDeviceInfo(i);
         if (!info) continue;
         if (!info->output) continue;
         wxString interf = wxSafeConvertMB2WX(info->interf);
         wxString name = wxSafeConvertMB2WX(info->name);
         interf.Append(wxT(": ")).Append(name);
         if (wxStrcmp(interf, playbackDeviceName) == 0) {
            playbackDevice = i;
         }
      }
   } // (else playback device has Pm_GetDefaultOuputDeviceID())
 
   if (playbackDevice < 0)
      return false;
 
   /* open output device */
   mLastPmError = Pm_OpenOutput(&mMidiStream,
                                playbackDevice,
                                NULL,
                                0,
                                &::MidiTime,
                                this, // supplies pInfo argument to MidiTime
                                MIDI_MINIMAL_LATENCY_MS);
   if (mLastPmError == pmNoError) {
      mMidiStreamActive = true;
      mMidiPaused = false;
      mMidiLoopPasses = 0;
      mMidiOutputComplete = false;
      mMaxMidiTimestamp = 0;
      PrepareMidiIterator(true, mPlaybackSchedule.mT0, 0);
 
      // It is ok to call this now, but do not send timestamped midi
      // until after the first audio callback, which provides necessary
      // data for MidiTime().
      Pm_Synchronize(mMidiStream); // start using timestamps
   }
   return (mLastPmError == pmNoError);
}
 
void MIDIPlay::StopOtherStream()
{
   if (mMidiStream && mMidiStreamActive) {
      /* Stop Midi playback */
      mMidiStreamActive = false;
 
      mMidiOutputComplete = true;
 
      // now we can assume "ownership" of the mMidiStream
      // if output in progress, send all off, etc.
      AllNotesOff();
      // AllNotesOff() should be sufficient to stop everything, but
      // in Linux, if you Pm_Close() immediately, it looks like
      // messages are dropped. ALSA then seems to send All Sound Off
      // and Reset All Controllers messages, but not all synthesizers
      // respond to these messages. This is probably a bug in PortMidi
      // if the All Off messages do not get out, but for security,
      // delay a bit so that messages can be delivered before closing
      // the stream. Add 2ms of "padding" to avoid any rounding errors.
      while (mMaxMidiTimestamp + 2 > MidiTime()) {
         using namespace std::chrono;
         std::this_thread::sleep_for(1ms); // deliver the all-off messages
      }
      Pm_Close(mMidiStream);
      mMidiStream = NULL;
      mIterator.reset();
 
      // set in_use flags to false
      int nTracks = mMidiPlaybackTracks.size();
      for (int i = 0; i < nTracks; i++) {
         const auto t = mMidiPlaybackTracks[i].get();
         Alg_seq_ptr seq = &t->GetSeq();
         seq->set_in_use(false);
      }
   }
 
   mMidiPlaybackTracks.clear();
}
 
double Iterator::UncorrectedMidiEventTime(double pauseTime)
{
   double time;
   if (mPlaybackSchedule.mEnvelope)
      time =
         mPlaybackSchedule.RealDuration(
            GetNextEventTime() - mMIDIPlay.MidiLoopOffset())
         + mPlaybackSchedule.mT0 +
           (mMIDIPlay.mMidiLoopPasses * mPlaybackSchedule.mWarpedLength);
   else
      time = GetNextEventTime();
 
   return time + pauseTime;
}
 
bool Iterator::Unmuted(bool hasSolo) const
{
   int channel = (mNextEvent->chan) & 0xF; // must be in [0..15]
   if (!mNextEventTrack->IsVisibleChan(channel))
      return false;
   const bool channelIsMute = hasSolo
      ? !mNextEventTrack->GetSolo()
      : mNextEventTrack->GetMute();
   return !channelIsMute;
}
 
bool Iterator::OutputEvent(double pauseTime, bool midiStateOnly, bool hasSolo)
{
   int channel = (mNextEvent->chan) & 0xF; // must be in [0..15]
   int command = -1;
   int data1 = -1;
   int data2 = -1;
 
   double eventTime = UncorrectedMidiEventTime(pauseTime);
 
   // 0.0005 is for rounding
   double time = eventTime + 0.0005 -
                 (mMIDIPlay.mSynthLatency * 0.001);
 
   time += 1; // MidiTime() has a 1s offset
   // state changes have to go out without delay because the
   // midi stream time gets reset when playback starts, and
   // we don't want to leave any control changes scheduled for later
   if (time < 0 || midiStateOnly)
      time = 0;
   PmTimestamp timestamp = (PmTimestamp) (time * 1000); /* s to ms */
 
   // The special event gAllNotesOff means "end of playback, send
   // all notes off on all channels"
   if (mNextEvent == &gAllNotesOff) {
      bool looping = mPlaybackSchedule.GetPolicy().Looping(mPlaybackSchedule);
      mMIDIPlay.AllNotesOff(looping);
      return true;
   }
 
   // (RBD)
   // if mNextEvent's channel is visible, play it, visibility can
   // be updated while playing.
   
   // Be careful: if we have a note-off,
   // then we must not pay attention to the channel selection
   // or mute/solo buttons because we must turn the note off
   // even if the user changed something after the note began.
 
   // Note that because multiple tracks can output to the same
   // MIDI channels, it is not a good idea to send "All Notes Off"
   // when the user presses the mute button. We have no easy way
   // to know what notes are sounding on any given muted track, so
   // we'll just wait for the note-off events to happen.
 
   // (PRL)
   // Does that mean, try to get right results when playing to the same SET
   // of MIDI channels, but tracks play to different channels?  In the
   // case that two unrelated tracks try to merge events, for notes that
   // overlap in time, to the same channel -- then one track still turns off
   // a note that another turned on.  Maybe the prevention of this case belongs
   // to higher levels of the program, and should just be assumed here.
 
   // (RBD)
   // Also note that note-offs are only sent when we call
   // mIterator->request_note_off(), so notes that are not played
   // will not generate random note-offs. There is the interesting
   // case that if the playback is paused, all-notes-off WILL be sent
   // and if playback resumes, the pending note-off events WILL also
   // be sent (but if that is a problem, there would also be a problem
   // in the non-pause case.
   const bool sendIt = [&]{
      const bool isNote = mNextEvent->is_note();
      if (!(isNote && mNextIsNoteOn))
         // Regardless of channel visibility state,
         // always send note-off events,
         // and update events (program change, control change, pressure, bend)
         // in case the user changes the muting during play
         return true;
      return Unmuted(hasSolo);
   }();
   if (sendIt) {
      // Note event
      if (mNextEvent->is_note() && !midiStateOnly) {
         // Pitch and velocity
         data1 = mNextEvent->get_pitch();
         if (mNextIsNoteOn) {
            data2 = mNextEvent->get_loud(); // get velocity
            int offset = mNextEventTrack->GetVelocity();
            data2 += offset; // offset comes from per-track slider
            // clip velocity to insure a legal note-on value
            data2 = (data2 < 1 ? 1 : (data2 > 127 ? 127 : data2));
            // since we are going to play this note, we need to get a note_off
            it.request_note_off();
 
#ifdef AUDIO_IO_GB_MIDI_WORKAROUND
            mMIDIPlay.mPendingNotesOff.push_back(std::make_pair(channel, data1));
#endif
         }
         else {
            data2 = 0; // 0 velocity means "note off"
#ifdef AUDIO_IO_GB_MIDI_WORKAROUND
            auto end = mMIDIPlay.mPendingNotesOff.end();
            auto iter = std::find(
               mMIDIPlay.mPendingNotesOff.begin(), end, std::make_pair(channel, data1) );
            if (iter != end)
               mMIDIPlay.mPendingNotesOff.erase(iter);
#endif
         }
         command = 0x90; // MIDI NOTE ON (or OFF when velocity == 0)
      // Update event
      } else if (mNextEvent->is_update()) {
         // this code is based on allegrosmfwr.cpp -- it could be improved
         // by comparing attribute pointers instead of string compares
         Alg_update_ptr update = static_cast<Alg_update_ptr>(mNextEvent);
         const char *name = update->get_attribute();
 
         if (!strcmp(name, "programi")) {
            // Instrument change
            data1 = update->parameter.i;
            data2 = 0;
            command = 0xC0; // MIDI PROGRAM CHANGE
         } else if (!strncmp(name, "control", 7)) {
            // Controller change
 
            // The number of the controller being changed is embedded
            // in the parameter name.
            data1 = atoi(name + 7);
            // Allegro normalizes controller values
            data2 = ROUND(update->parameter.r * 127);
            command = 0xB0;
         } else if (!strcmp(name, "bendr")) {
            // Bend change
 
            // Reverse Allegro's post-processing of bend values
            int temp = ROUND(0x2000 * (update->parameter.r + 1));
            if (temp > 0x3fff) temp = 0x3fff; // 14 bits maximum
            if (temp < 0) temp = 0;
            data1 = temp & 0x7f; // low 7 bits
            data2 = temp >> 7;   // high 7 bits
            command = 0xE0; // MIDI PITCH BEND
         } else if (!strcmp(name, "pressurer")) {
            // Pressure change
            data1 = (int) (update->parameter.r * 127);
            if (update->get_identifier() < 0) {
               // Channel pressure
               data2 = 0;
               command = 0xD0; // MIDI CHANNEL PRESSURE
            } else {
               // Key pressure
               data2 = data1;
               data1 = update->get_identifier();
               command = 0xA0; // MIDI POLY PRESSURE
            }
         }
      }
      if (command != -1) {
         // keep track of greatest timestamp used
         if (timestamp > mMIDIPlay.mMaxMidiTimestamp) {
            mMIDIPlay.mMaxMidiTimestamp = timestamp;
         }
         Pm_WriteShort(mMIDIPlay.mMidiStream, timestamp,
                    Pm_Message((int) (command + channel),
                                  (long) data1, (long) data2));
         /* wxPrintf("Pm_WriteShort %lx (%p) @ %d, advance %d\n",
                Pm_Message((int) (command + channel),
                           (long) data1, (long) data2),
                           mNextEvent, timestamp, timestamp - Pt_Time()); */
      }
   }
   return false;
}
 
void Iterator::GetNextEvent()
{
   mNextEventTrack = nullptr; // clear it just to be safe
   // now get the next event and the track from which it came
   double nextOffset;
   auto midiLoopOffset = mMIDIPlay.MidiLoopOffset();
   mNextEvent = it.next(&mNextIsNoteOn,
      // Allegro retrieves the "cookie" for the event, which is a NoteTrack
      reinterpret_cast<void **>(&mNextEventTrack),
      &nextOffset, mPlaybackSchedule.mT1 + midiLoopOffset);
 
   mNextEventTime  = mPlaybackSchedule.mT1 + midiLoopOffset + 1;
   if (mNextEvent) {
      mNextEventTime = (mNextIsNoteOn ? mNextEvent->time :
                              mNextEvent->get_end_time()) + nextOffset;;
   }
   if (mNextEventTime > (mPlaybackSchedule.mT1 + midiLoopOffset)){ // terminate playback at mT1
      mNextEvent = &gAllNotesOff;
      mNextEventTime = mPlaybackSchedule.mT1 + midiLoopOffset;
      mNextIsNoteOn = true; // do not look at duration
   }
}
 
void MIDIPlay::FillOtherBuffers(
   double rate, unsigned long pauseFrames, bool paused, bool hasSolo)
{
   if (!mMidiStream)
      return;
 
   // If not paused, fill buffers.
   if (paused)
      return;
 
   // If we compute until GetNextEventTime() > current audio time,
   // we would have a built-in compute-ahead of mAudioOutLatency, and
   // it's probably good to compute MIDI when we compute audio (so when
   // we stop, both stop about the same time).
   double time = AudioTime(rate); // compute to here
   // But if mAudioOutLatency is very low, we might need some extra
   // compute-ahead to deal with mSynthLatency or even this thread.
   double actual_latency  = (MIDI_MINIMAL_LATENCY_MS + mSynthLatency) * 0.001;
   if (actual_latency > mAudioOutLatency) {
       time += actual_latency - mAudioOutLatency;
   }
   while (mIterator &&
          mIterator->mNextEvent &&
          mIterator->UncorrectedMidiEventTime(PauseTime(rate, pauseFrames)) < time) {
      if (mIterator->OutputEvent(PauseTime(rate, pauseFrames), false, hasSolo)) {
         if (mPlaybackSchedule.GetPolicy().Looping(mPlaybackSchedule)) {
            // jump back to beginning of loop
            ++mMidiLoopPasses;
            PrepareMidiIterator(false, mPlaybackSchedule.mT0, MidiLoopOffset());
         }
         else
            mIterator.reset();
      }
      else if (mIterator)
         mIterator->GetNextEvent();
   }
}
 
double MIDIPlay::PauseTime(double rate, unsigned long pauseFrames)
{
   return pauseFrames / rate;
}
 
 
// MidiTime() is an estimate in milliseconds of the current audio
// output (DAC) time + 1s. In other words, what audacity track time
// corresponds to the audio (including pause insertions) at the output?
//
PmTimestamp MIDIPlay::MidiTime()
{
   // note: the extra 0.0005 is for rounding. Round down by casting to
   // unsigned long, then convert to PmTimeStamp (currently signed)
 
   // PRL: the time correction is really Midi latency achieved by different
   // means than specifying it to Pm_OpenStream.  The use of the accumulated
   // sample count generated by the audio callback (in AudioTime()) might also
   // have the virtue of keeping the Midi output synched with audio.
 
   PmTimestamp ts;
   // subtract latency here because mSystemMinusAudioTime gets us
   // to the current *write* time, but we're writing ahead by audio output
   // latency (mAudioOutLatency).
   double now = SystemTime(mUsingAlsa);
   ts = (PmTimestamp) ((unsigned long)
         (1000 * (now + 1.0005 -
                  mSystemMinusAudioTimePlusLatency)));
   // wxPrintf("AudioIO::MidiTime() %d time %g sys-aud %g\n",
   //        ts, now, mSystemMinusAudioTime);
   return ts + MIDI_MINIMAL_LATENCY_MS;
}
 
 
void MIDIPlay::AllNotesOff(bool looping)
{
#ifdef __WXGTK__
   bool doDelay = !looping;
#else
   bool doDelay = false;
   static_cast<void>(looping);// compiler food.
#endif
 
   // to keep track of when MIDI should all be delivered,
   // update mMaxMidiTimestamp to now:
   PmTimestamp now = MidiTime();
   if (mMaxMidiTimestamp < now) {
       mMaxMidiTimestamp = now;
   }
#ifdef AUDIO_IO_GB_MIDI_WORKAROUND
   // PRL:
   // Send individual note-off messages for each note-on not yet paired.
 
   // RBD:
   // Even this did not work as planned. My guess is ALSA does not use
   // a "stable sort" for timed messages, so that when a note-off is
   // added later at the same time as a future note-on, the order is
   // not respected, and the note-off can go first, leaving a stuck note.
   // The workaround here is to use mMaxMidiTimestamp to ensure that
   // note-offs come at least 1ms later than any previous message
 
   // PRL:
   // I think we should do that only when stopping or pausing, not when looping
   // Note that on Linux, MIDI always uses ALSA, no matter whether portaudio
   // uses some other host api.
 
   mMaxMidiTimestamp += 1;
   for (const auto &pair : mPendingNotesOff) {
      Pm_WriteShort(mMidiStream,
                    (doDelay ? mMaxMidiTimestamp : 0),
                    Pm_Message(
         0x90 + pair.first, pair.second, 0));
      mMaxMidiTimestamp++; // allow 1ms per note-off
   }
   mPendingNotesOff.clear();
 
   // Proceed to do the usual messages too.
#endif
 
   for (int chan = 0; chan < 16; chan++) {
      Pm_WriteShort(mMidiStream,
                    (doDelay ? mMaxMidiTimestamp : 0),
                    Pm_Message(0xB0 + chan, 0x7B, 0));
      mMaxMidiTimestamp++; // allow 1ms per all-notes-off
   }
}
 
void MIDIPlay::ComputeOtherTimings(double rate, bool paused,
   const PaStreamCallbackTimeInfo *timeInfo,
   unsigned long framesPerBuffer
   )
{
   if (mCallbackCount++ == 0) {
       // This is effectively mSystemMinusAudioTime when the buffer is empty:
       mStartTime = SystemTime(mUsingAlsa) - mPlaybackSchedule.mT0;
       // later, mStartTime - mSystemMinusAudioTime will tell us latency
   }
 
   /* for Linux, estimate a smooth audio time as a slowly-changing
      offset from system time */
   // rnow is system time as a double to simplify math
   double rnow = SystemTime(mUsingAlsa);
   // anow is next-sample-to-be-computed audio time as a double
   double anow = AudioTime(rate);
 
   if (mUsingAlsa) {
      // timeInfo's fields are not all reliable.
 
      // enow is audio time estimated from our clock synchronization protocol,
      //   which produces mSystemMinusAudioTime. But we want the estimate
      //   to drift low, so we steadily increase mSystemMinusAudioTime to
      //   simulate a fast system clock or a slow audio clock. If anow > enow,
      //   we'll update mSystemMinusAudioTime to keep in sync. (You might think
      //   we could just use anow as the "truth", but it has a lot of jitter,
      //   so we are using enow to smooth out this jitter, in fact to < 1ms.)
      // Add worst-case clock drift using previous framesPerBuffer:
      const auto increase =
         mAudioFramesPerBuffer * 0.0002 / rate;
      mSystemMinusAudioTime += increase;
      mSystemMinusAudioTimePlusLatency += increase;
      double enow = rnow - mSystemMinusAudioTime;
 
 
      // now, use anow instead if it is ahead of enow
      if (anow > enow) {
         mSystemMinusAudioTime = rnow - anow;
         // Update our mAudioOutLatency estimate during the first 20 callbacks.
         // During this period, the buffer should fill. Once we have a good
         // estimate of mSystemMinusAudioTime (expected in fewer than 20 callbacks)
         // we want to stop the updating in case there is clock drift, which would
         // cause the mAudioOutLatency estimation to drift as well. The clock drift
         // in the first 20 callbacks should be negligible, however.
         if (mCallbackCount < 20) {
            mAudioOutLatency = mStartTime -
               mSystemMinusAudioTime;
         }
         mSystemMinusAudioTimePlusLatency =
            mSystemMinusAudioTime + mAudioOutLatency;
      }
   }
   else {
      // If not using Alsa, rely on timeInfo to have meaningful values that are
      // more precise than the output latency value reported at stream start.
      mSystemMinusAudioTime = rnow - anow;
      mSystemMinusAudioTimePlusLatency =
         mSystemMinusAudioTime +
            (timeInfo->outputBufferDacTime - timeInfo->currentTime);
   }
 
   mAudioFramesPerBuffer = framesPerBuffer;
   mNumFrames += framesPerBuffer;
 
   // Keep track of time paused.
   if (paused) {
      if (!mMidiPaused) {
         mMidiPaused = true;
         AllNotesOff(); // to avoid hanging notes during pause
      }
   }
   else if (mMidiPaused)
      mMidiPaused = false;
}
 
unsigned MIDIPlay::CountOtherSoloTracks() const
{
   return std::count_if(
      mMidiPlaybackTracks.begin(), mMidiPlaybackTracks.end(),
      [](const auto &pTrack){ return pTrack->GetSolo(); } );
}
 
void MIDIPlay::SignalOtherCompletion()
{
   mMidiOutputComplete = true;
}
}
 
bool MIDIPlay::IsActive()
{
   return ( mMidiStreamActive && !mMidiOutputComplete );
}
 
bool MIDIPlay::IsOtherStreamActive() const
{
   return IsActive();
}
 
AudioIODiagnostics MIDIPlay::Dump() const
{
   return {
      wxT("mididev.txt"),
      GetMIDIDeviceInfo(),
      wxT("MIDI Device Info")
   };
}