bool LOS::importMidi(const QString name, bool merge)/*{{{*/
{
bool popenFlag;
FILE* fp = fileOpen(this, name, QString(".mid"), "r", popenFlag);
if (fp == 0)
return true;
MidiFile mf(fp);
bool rv = mf.read();
popenFlag ? pclose(fp) : fclose(fp);
if (rv)
{
QString s(tr("reading midifile\n "));
s += name;
s += tr("\nfailed: ");
s += mf.error();
QMessageBox::critical(this, QString("LOS"), s);
return rv;
}
//
// evaluate song Type (GM, XG, GS, unknown)
//
MType t = song->mtype();
if (!merge)
{
t = mf.mtype();
song->setMType(t);
}
MidiInstrument* instr = 0;
for (iMidiInstrument i = midiInstruments.begin(); i != midiInstruments.end(); ++i)
{
MidiInstrument* mi = *i;
if ((mi->iname() == "GM" && ((t == MT_UNKNOWN) || (t == MT_GM)))
|| ((mi->iname() == "GS") && (t == MT_GS))
|| ((mi->iname() == "XG") && (t == MT_XG)))
{
instr = mi;
break;
}
}
if (instr == 0)
{
// the standard instrument files (GM, GS, XG) must be present
printf("no instrument, type %d\n", t);
return true;
//abort();
}
MidiFileTrackList* etl = mf.trackList();
int division = mf.division();
//
// create MidiTrack and copy events to ->events()
// - combine note on/off events
// - calculate tick value for internal resolution
//
int mPort = getFreeMidiPort();
for (iMidiFileTrack t = etl->begin(); t != etl->end(); ++t)
{
MPEventList* el = &((*t)->events);
if (el->empty())
continue;
//
// if we split the track, SYSEX and META events go into
// the first target track
bool first = true;
// somewhat silly and slooow:
QList<QPair<int, int> > eventChannelList;
if(mPort >= 0 && mPort < MIDI_PORTS)
{
for (int channel = 0; channel < MIDI_CHANNELS; ++channel)
{
//
// check if there are any events for port/channel in track:
//
iMPEvent i;
for (i = el->begin(); i != el->end(); ++i)
{
MidiPlayEvent ev = *i;
if (ev.type() != ME_SYSEX && ev.type() != ME_META && ev.channel() == channel)
break;
}
if (i == el->end())
continue;
MidiTrack* track = new MidiTrack();
track->setDefaultName();
track->setMasterFlag(true);
track->setOutChannel(channel);
track->setOutPort(mPort);
MidiPort* mport = &midiPorts[track->outPort()];
// this overwrites any instrument set for this port:
mport->setInstrument(instr);
EventList* mel = track->events();
buildMidiEventList(mel, el, track, division, first, false, false);
first = false;
processTrack(track);
song->insertTrack(track, -1);
//Create the Audio input side of the track
Track* input = song->addTrackByName(QString("i").append(track->name()), Track::AUDIO_INPUT, -1, false, false);
if(input)
{
input->setMasterFlag(false);
input->setChainMaster(track->id());
track->addManagedTrack(input->id());
}
}
}
if (first)
{
//
// track does only contain non-channel messages
// (SYSEX or META)
//
MidiTrack* track = new MidiTrack();
track->setDefaultName();
track->setMasterFlag(true);
track->setOutChannel(0);
track->setOutPort(mPort);
EventList* mel = track->events();
//buildMidiEventList(mel, el, track, division, true);
// Do SysexMeta. Don't do loops.
buildMidiEventList(mel, el, track, division, true, false, false);
processTrack(track);
song->insertTrack(track, -1);
//Create the Audio input side of the track
Track* input = song->addTrackByName(QString("i").append(track->name()), Track::AUDIO_INPUT, -1, false, false);
if(input)
{
input->setMasterFlag(false);
input->setChainMaster(track->id());
track->addManagedTrack(input->id());
}
}
mPort++;
//FIXME: Provice a non-iterative way to do this using the new losMidiPorts hash
//Or maintain a list of configured or inuse ports
while((&midiPorts[mPort])->device() && mPort < MIDI_PORTS)
mPort++;//Just incase we have a configured port after an empty one
}
if (!merge)
{
TrackList* tl = song->tracks();
if (!tl->empty())
{
Track* track = tl->front();
track->setSelected(true);
}
song->initLen();
int z, n;
AL::sigmap.timesig(0, z, n);
int tempo = tempomap.tempo(0);
transport->setTimesig(z, n);
transport->setTempo(tempo);
bool masterF = !tempomap.empty();
song->setMasterFlag(masterF);
transport->setMasterFlag(masterF);
song->updatePos();
composer->reset();
}
else
{
song->initLen();
}
return false;
}/*}}}*/
void Song::processAutomations(const TrackList &tracklist, MidiTime timeStart, fpp_t)
{
AutomatedValueMap values;
QSet<const AutomatableModel*> recordedModels;
TrackContainer* container = this;
int tcoNum = -1;
switch (m_playMode)
{
case Mode_PlaySong:
break;
case Mode_PlayBB:
{
Q_ASSERT(tracklist.size() == 1);
Q_ASSERT(tracklist.at(0)->type() == Track::BBTrack);
auto bbTrack = dynamic_cast<BBTrack*>(tracklist.at(0));
auto bbContainer = Engine::getBBTrackContainer();
container = bbContainer;
tcoNum = bbTrack->index();
}
break;
default:
return;
}
values = container->automatedValuesAt(timeStart, tcoNum);
TrackList tracks = container->tracks();
Track::tcoVector tcos;
for (Track* track : tracks)
{
if (track->type() == Track::AutomationTrack) {
track->getTCOsInRange(tcos, 0, timeStart);
}
}
// Process recording
for (TrackContentObject* tco : tcos)
{
auto p = dynamic_cast<AutomationPattern *>(tco);
MidiTime relTime = timeStart - p->startPosition();
if (p->isRecording() && relTime >= 0 && relTime < p->length())
{
const AutomatableModel* recordedModel = p->firstObject();
p->recordValue(relTime, recordedModel->value<float>());
recordedModels << recordedModel;
}
}
// Apply values
for (auto it = values.begin(); it != values.end(); it++)
{
if (! recordedModels.contains(it.key()))
{
it.key()->setAutomatedValue(it.value());
}
}
}
bool ExportCL(AudacityProject *project, bool stereo, wxString fName,
bool selectionOnly, double t0, double t1, MixerSpec *mixerSpec)
{
int rate = int(project->GetRate() + 0.5);
wxWindow *parent = project;
TrackList *tracks = project->GetTracks();
wxString command = gPrefs->Read(wxT("/FileFormats/ExternalProgramExportCommand"), wxT("lame - '%f'"));
command.Replace(wxT("%f"), fName);
/* establish parameters */
int channels = stereo ? 2 : 1;
unsigned long totalSamples = (unsigned long)((t1 - t0) * rate + 0.5);
unsigned long sampleBytes = totalSamples * channels * SAMPLE_SIZE(int16Sample);
/* fill up the wav header */
wav_header header;
header.riffID[0] = 'R';
header.riffID[1] = 'I';
header.riffID[2] = 'F';
header.riffID[3] = 'F';
header.riffType[0] = 'W';
header.riffType[1] = 'A';
header.riffType[2] = 'V';
header.riffType[3] = 'E';
header.lenAfterRiff = sampleBytes + 32;
header.fmtID[0] = 'f';
header.fmtID[1] = 'm';
header.fmtID[2] = 't';
header.fmtID[3] = ' ';
header.formatChunkLen = 16;
header.formatTag = 1;
header.channels = channels;
header.sampleRate = rate;
header.bitsPerSample = SAMPLE_SIZE(int16Sample) * 8;
header.blockAlign = header.bitsPerSample * header.channels;
header.avgBytesPerSec = header.sampleRate * header.blockAlign;
header.dataID[0] = 'd';
header.dataID[1] = 'a';
header.dataID[2] = 't';
header.dataID[3] = 'a';
header.dataLen = sampleBytes;
FILE *pipe = popen(OSFILENAME(command), "w");
/* write the header */
fwrite( &header, sizeof(wav_header), 1, pipe );
sampleCount maxBlockLen = 44100 * 5;
bool cancelling = false;
int numWaveTracks;
WaveTrack **waveTracks;
tracks->GetWaveTracks(selectionOnly, &numWaveTracks, &waveTracks);
Mixer *mixer = new Mixer(numWaveTracks, waveTracks,
tracks->GetTimeTrack(),
t0, t1,
channels, maxBlockLen, true,
rate, int16Sample, true, mixerSpec);
GetActiveProject()->ProgressShow(_("Export"),
selectionOnly ?
_("Exporting the selected audio using command-line encoder") :
_("Exporting the entire project using command-line encoder"));
while(!cancelling) {
sampleCount numSamples = mixer->Process(maxBlockLen);
if (numSamples == 0)
break;
samplePtr mixed = mixer->GetBuffer();
char *buffer = new char[numSamples * SAMPLE_SIZE(int16Sample) * channels];
wxASSERT(buffer);
// Byte-swapping is neccesary on big-endian machines, since
// WAV files are little-endian
#if wxBYTE_ORDER == wxBIG_ENDIAN
{
short *buffer = (short*)mixed;
for( int i = 0; i < numSamples; i++ )
buffer[i] = wxINT16_SWAP_ON_BE(buffer[i]);
}
#endif
fwrite( mixed, numSamples * channels * SAMPLE_SIZE(int16Sample), 1, pipe );
int progressvalue = int (1000 * ((mixer->MixGetCurrentTime()-t0) /
(t1-t0)));
cancelling = !GetActiveProject()->ProgressUpdate(progressvalue);
delete[]buffer;
}
GetActiveProject()->ProgressHide();
delete mixer;
pclose( pipe );
return true;
}
PyObject* getParts(PyObject*, PyObject* args)
{
TrackList* tracks = song->tracks();
const char* trackname;
if (!PyArg_ParseTuple(args, "s", &trackname))
{
return NULL;
}
PyObject* pyparts = Py_BuildValue("[]");
for (ciTrack t = tracks->begin(); t != tracks->end(); ++t)
{
Track* track = *t;
if (track->name() != trackname)
continue;
PartList* parts = track->parts();
for (ciPart p = parts->begin(); p != parts->end(); p++)
{
Part* part = p->second;
MidiPart* mpart = (MidiPart*) part;
PyObject* pypart = PyDict_New();
int tick = mpart->tick();
int lentick = mpart->lenTick();
int serialnr = mpart->sn();
PyObject* pstrtick = Py_BuildValue("s", "tick");
PyObject* pitick = Py_BuildValue("i", tick);
PyObject* pstrid = Py_BuildValue("s", "id");
PyObject* pstrserial = Py_BuildValue("i", serialnr);
PyObject* pstrlen = Py_BuildValue("s", "len");
PyObject* pstrtick2 = Py_BuildValue("i", lentick);
PyDict_SetItem(pypart, pstrtick, pitick);
PyDict_SetItem(pypart, pstrid, pstrserial);
PyDict_SetItem(pypart, pstrlen, pstrtick2);
Py_DECREF(pstrtick);
Py_DECREF(pitick);
Py_DECREF(pstrid);
Py_DECREF(pstrserial);
Py_DECREF(pstrlen);
Py_DECREF(pstrtick2);
// Pack midi events into list before wrapping it all up
EventList* events = mpart->events();
PyObject* pyevents = Py_BuildValue("[]");
for (ciEvent e = events->begin(); e != events->end(); e++)
{
PyObject* pyevent = PyDict_New(); // The event structure - a dictionary with keys 'type','tick','data'
const Event& event = e->second;
unsigned tick = e->first;
PyObject* eventdata = Py_BuildValue("[i,i,i]", event.dataA(), event.dataB(), event.dataC());
PyObject* pstrdata = Py_BuildValue("s", "data");
pstrtick = Py_BuildValue("s", "tick");
PyObject* pitickval = Py_BuildValue("i", tick);
PyDict_SetItem(pyevent, pstrdata, eventdata);
PyDict_SetItem(pyevent, pstrtick, pitickval);
Py_DECREF(eventdata);
Py_DECREF(pstrdata);
Py_DECREF(pstrtick);
Py_DECREF(pitickval);
switch (event.type())
{
case Note:
{
PyObject* pstrtype = Py_BuildValue("s", "type");
PyObject* pstrnote = Py_BuildValue("s", "note");
PyObject* pstrlen = Py_BuildValue("s", "len");
PyObject* pilentick = Py_BuildValue("i", event.lenTick());
PyDict_SetItem(pyevent, pstrtype, pstrnote);
PyDict_SetItem(pyevent, pstrlen, pilentick);
Py_DECREF(pstrtype);
Py_DECREF(pstrnote);
Py_DECREF(pstrlen);
Py_DECREF(pilentick);
break;
}
case Controller:
{
PyObject* pstrtype = Py_BuildValue("s", "type");
PyObject* pstrctrl = Py_BuildValue("s", "ctrl");
PyDict_SetItem(pyevent, pstrtype, pstrctrl);
Py_DECREF(pstrtype);
Py_DECREF(pstrctrl);
break;
}
default:
printf("Event type not supported yet: %d\n", event.type());
break;
}
PyList_Append(pyevents, pyevent);
Py_DECREF(pyevent);
}
Py_DECREF(pyevents);
// Add the event list to the pypart dictionary
PyObject* pystrevents = Py_BuildValue("s", "events");
PyDict_SetItem(pypart, pystrevents, pyevents);
Py_DECREF(pystrevents);
PyList_Append(pyparts, pypart);
Py_DECREF(pypart);
}
return pyparts;
}
return NULL;
}
void Song::processNextBuffer()
{
m_vstSyncController.setPlaybackJumped( false );
// if not playing, nothing to do
if( m_playing == false )
{
return;
}
TrackList trackList;
int tcoNum = -1; // track content object number
// determine the list of tracks to play and the track content object
// (TCO) number
switch( m_playMode )
{
case Mode_PlaySong:
trackList = tracks();
// at song-start we have to reset the LFOs
if( m_playPos[Mode_PlaySong] == 0 )
{
EnvelopeAndLfoParameters::instances()->reset();
}
break;
case Mode_PlayBB:
if( Engine::getBBTrackContainer()->numOfBBs() > 0 )
{
tcoNum = Engine::getBBTrackContainer()->
currentBB();
trackList.push_back( BBTrack::findBBTrack(
tcoNum ) );
}
break;
case Mode_PlayPattern:
if( m_patternToPlay != NULL )
{
tcoNum = m_patternToPlay->getTrack()->
getTCONum( m_patternToPlay );
trackList.push_back(
m_patternToPlay->getTrack() );
}
break;
default:
return;
}
// if we have no tracks to play, nothing to do
if( trackList.empty() == true )
{
return;
}
// check for looping-mode and act if necessary
TimeLineWidget * tl = m_playPos[m_playMode].m_timeLine;
bool checkLoop =
tl != NULL && m_exporting == false && tl->loopPointsEnabled();
if( checkLoop )
{
// if looping-mode is enabled and we are outside of the looping
// range, go to the beginning of the range
if( m_playPos[m_playMode] < tl->loopBegin() ||
m_playPos[m_playMode] >= tl->loopEnd() )
{
setToTime(tl->loopBegin());
m_playPos[m_playMode].setTicks(
tl->loopBegin().getTicks() );
m_vstSyncController.setPlaybackJumped( true );
emit updateSampleTracks();
}
}
if( m_playPos[m_playMode].jumped() )
{
m_vstSyncController.setPlaybackJumped( true );
m_playPos[m_playMode].setJumped( false );
}
f_cnt_t framesPlayed = 0;
const float framesPerTick = Engine::framesPerTick();
while( framesPlayed < Engine::mixer()->framesPerPeriod() )
{
m_vstSyncController.update();
float currentFrame = m_playPos[m_playMode].currentFrame();
// did we play a tick?
if( currentFrame >= framesPerTick )
{
int ticks = m_playPos[m_playMode].getTicks() +
( int )( currentFrame / framesPerTick );
// did we play a whole tact?
if( ticks >= MidiTime::ticksPerTact() )
{
// per default we just continue playing even if
// there's no more stuff to play
// (song-play-mode)
int maxTact = m_playPos[m_playMode].getTact()
+ 2;
// then decide whether to go over to next tact
// or to loop back to first tact
if( m_playMode == Mode_PlayBB )
{
maxTact = Engine::getBBTrackContainer()
->lengthOfCurrentBB();
}
else if( m_playMode == Mode_PlayPattern &&
m_loopPattern == true &&
tl != NULL &&
tl->loopPointsEnabled() == false )
{
maxTact = m_patternToPlay->length()
.getTact();
}
// end of played object reached?
if( m_playPos[m_playMode].getTact() + 1
>= maxTact )
{
// then start from beginning and keep
// offset
ticks %= ( maxTact * MidiTime::ticksPerTact() );
// wrap milli second counter
setToTimeByTicks(ticks);
m_vstSyncController.setPlaybackJumped( true );
}
}
m_playPos[m_playMode].setTicks( ticks );
if (checkLoop || m_loopRenderRemaining > 1)
{
m_vstSyncController.startCycle(
tl->loopBegin().getTicks(), tl->loopEnd().getTicks() );
// if looping-mode is enabled and we have got
// past the looping range, return to the
// beginning of the range
if( m_playPos[m_playMode] >= tl->loopEnd() )
{
if (m_loopRenderRemaining > 1)
m_loopRenderRemaining--;
ticks = tl->loopBegin().getTicks();
m_playPos[m_playMode].setTicks( ticks );
setToTime(tl->loopBegin());
m_vstSyncController.setPlaybackJumped( true );
emit updateSampleTracks();
}
}
else
{
m_vstSyncController.stopCycle();
}
currentFrame = fmodf( currentFrame, framesPerTick );
m_playPos[m_playMode].setCurrentFrame( currentFrame );
}
if( framesPlayed == 0 )
{
// update VST sync position after we've corrected frame/
// tick count but before actually playing any frames
m_vstSyncController.setAbsolutePosition(
m_playPos[m_playMode].getTicks()
+ m_playPos[m_playMode].currentFrame()
/ (double) framesPerTick );
}
f_cnt_t framesToPlay =
Engine::mixer()->framesPerPeriod() - framesPlayed;
f_cnt_t framesLeft = ( f_cnt_t )framesPerTick -
( f_cnt_t )currentFrame;
// skip last frame fraction
if( framesLeft == 0 )
{
++framesPlayed;
m_playPos[m_playMode].setCurrentFrame( currentFrame
+ 1.0f );
continue;
}
// do we have samples left in this tick but these are less
// than samples we have to play?
if( framesLeft < framesToPlay )
{
// then set framesToPlay to remaining samples, the
// rest will be played in next loop
framesToPlay = framesLeft;
}
if( ( f_cnt_t ) currentFrame == 0 )
{
processAutomations(trackList, m_playPos[m_playMode], framesToPlay);
// loop through all tracks and play them
for( int i = 0; i < trackList.size(); ++i )
{
trackList[i]->play( m_playPos[m_playMode],
framesToPlay,
framesPlayed, tcoNum );
}
}
// update frame-counters
framesPlayed += framesToPlay;
m_playPos[m_playMode].setCurrentFrame( framesToPlay +
currentFrame );
m_elapsedMilliSeconds[m_playMode] += MidiTime::ticksToMilliseconds(framesToPlay / framesPerTick, getTempo());
m_elapsedTacts = m_playPos[Mode_PlaySong].getTact();
m_elapsedTicks = ( m_playPos[Mode_PlaySong].getTicks() % ticksPerTact() ) / 48;
}
}
void ControlToolBar::OnRecord(wxCommandEvent &evt)
{
if (gAudioIO->IsBusy()) {
#if (AUDACITY_BRANDING == BRAND_AUDIOTOUCH)
if (mIsLocked) // Stop only if in locked mode.
{
this->StopPlaying(); // Stop recording.
mRecord->PopUp();
}
else
{
//v evt.GetClassInfo();
}
#else
mRecord->PopUp();
#endif
return;
}
mPlay->Disable();
#if (AUDACITY_BRANDING == BRAND_THINKLABS)
mLoopPlay->Disable();
#endif
#if (AUDACITY_BRANDING == BRAND_AUDIOTOUCH)
mStop->SetEnabled(mIsLocked);
#else
mStop->Enable();
#endif
mRewind->Disable();
mFF->Disable();
#if (AUDACITY_BRANDING == BRAND_AUDIOTOUCH)
mPause->SetEnabled(mIsLocked);
this->EnablePauseCommand(mIsLocked);
this->SetRecord(mIsLocked); // If locked, push Record down, else up.
#else
mPause->Enable();
mRecord->PushDown();
#endif
AudacityProject *p = GetActiveProject();
if (p) {
TrackList *t = p->GetTracks();
// Don't do this for Audiotouch. Not necessary. The performance hit is in the track being recorded, below.
#if (AUDACITY_BRANDING == BRAND_THINKLABS)
// For versions that default to dual wave/spectrum display,
// switch all tracks to WaveformDisplay on Record, for performance.
TrackListIterator iter(t);
for (Track* pTrack = iter.First(); pTrack; pTrack = iter.Next())
if (pTrack->GetKind() == Track::Wave)
((WaveTrack*)pTrack)->SetDisplay(WaveTrack::WaveformDisplay);
#endif
double t0 = p->GetSel0();
double t1 = p->GetSel1();
if (t1 == t0)
t1 = 1000000000.0; // record for a long, long time (tens of years)
#if (AUDACITY_BRANDING == BRAND_AUDIOTOUCH)
if (p->m_bWantAppendRecording)
{
t0 = t->GetEndTime();
t1 = 1000000000.0; // record for a long, long time (tens of years)
}
gAudioIO->SetWantLatencyCorrection(!p->m_bWantAppendRecording);
#endif
/* TODO: set up stereo tracks if that is how the user has set up
* their preferences, and choose sample format based on prefs */
WaveTrackArray newRecordingTracks, playbackTracks;
bool duplex;
gPrefs->Read("/AudioIO/Duplex", &duplex, true);
int recordingChannels = gPrefs->Read("/AudioIO/RecordChannels", 1);
if( duplex )
playbackTracks = t->GetWaveTrackArray(false);
else
playbackTracks = WaveTrackArray();
for( int c = 0; c < recordingChannels; c++ )
{
WaveTrack *newTrack = p->GetTrackFactory()->NewWaveTrack();
newTrack->SetOffset(t0);
newTrack->SetRate(p->GetRate());
//v for performance newTrack->SetDisplay(WaveTrack::WaveformAndSpectrumDisplay);
newTrack->SetDisplay(WaveTrack::WaveformDisplay);
if( recordingChannels == 2 )
{
if( c == 0 )
{
newTrack->SetChannel(Track::LeftChannel);
newTrack->SetLinked(true);
}
else
newTrack->SetChannel(Track::RightChannel);
}
else
{
newTrack->SetChannel( Track::MonoChannel );
}
newRecordingTracks.Add(newTrack);
}
int token = gAudioIO->StartStream(playbackTracks,
newRecordingTracks, t->GetTimeTrack(),
p->GetRate(), t0, t1);
bool success = (token != 0);
for( unsigned int i = 0; i < newRecordingTracks.GetCount(); i++ )
if (success)
t->Add(newRecordingTracks[i]);
else
delete newRecordingTracks[i];
if (success) {
p->SetAudioIOToken(token);
mBusyProject = p;
SetVUMeters(p);
#if (AUDACITY_BRANDING == BRAND_AUDIOTOUCH)
this->SetBackgroundColour(*wxRED); // red for Recording
this->Refresh();
#endif
}
else {
// msmeyer: Show error message if stream could not be opened
wxMessageBox(_("Error while opening sound device. Please check the input "
"device settings and the project sample rate."),
_("Error"), wxOK | wxICON_EXCLAMATION, this);
SetPlay(false);
#if (AUDACITY_BRANDING == BRAND_AUDIOTOUCH)
mStop->Enable(); // In case it was disabled above, based on mIsLocked.
#endif
SetStop(false);
SetRecord(false);
}
#if (AUDACITY_BRANDING == BRAND_AUDIOTOUCH)
p->OnZoomFitV();
#endif
}
}
void ControlToolBar::PlayPlayRegion(double t0, double t1,
bool looped /* = false */)
{
if (gAudioIO->IsBusy()) {
mPlay->PopUp();
#if (AUDACITY_BRANDING == BRAND_THINKLABS)
mLoopPlay->PopUp();
#elif (AUDACITY_BRANDING == BRAND_AUDIOTOUCH)
mPlay->Show();
mPause->Hide();
this->EnablePauseCommand(false);
#endif
return;
}
mStop->Enable();
mRewind->Disable();
mRecord->Disable();
mFF->Disable();
mPause->Enable();
this->EnablePauseCommand(true);
AudacityProject *p = GetActiveProject();
if (p) {
TrackList *t = p->GetTracks();
double maxofmins,minofmaxs;
// JS: clarified how the final play region is computed;
if (t1 == t0) {
// msmeyer: When playing looped, we play the whole file, if
// no range is selected. Otherwise, we play from t0 to end
if (looped) {
// msmeyer: always play from start
t0 = t->GetStartTime();
} else {
// move t0 to valid range
if (t0 < 0)
t0 = t->GetStartTime();
if (t0 > t->GetEndTime())
t0 = t->GetEndTime();
}
// always play to end
t1 = t->GetEndTime();
}
else {
// always t0 < t1 right?
// the set intersection between the play region and the
// valid range maximum of lower bounds
if (t0 < t->GetStartTime())
maxofmins = t->GetStartTime();
else
maxofmins = t0;
// minimum of upper bounds
if (t1 > t->GetEndTime())
minofmaxs = t->GetEndTime();
else
minofmaxs = t1;
// we test if the intersection has no volume
if (minofmaxs <= maxofmins) {
// no volume; play nothing
return;
}
else {
t0 = maxofmins;
t1 = minofmaxs;
}
}
bool success = false;
if (t1 > t0) {
int token =
gAudioIO->StartStream(t->GetWaveTrackArray(false),
WaveTrackArray(), t->GetTimeTrack(),
p->GetRate(), t0, t1, looped);
if (token != 0) {
success = true;
p->SetAudioIOToken(token);
mBusyProject = p;
SetVUMeters(p);
}
else {
// msmeyer: Show error message if stream could not be opened
wxMessageBox(_("Error while opening sound device. Please check the output "
"device settings and the project sample rate."),
_("Error"), wxOK | wxICON_EXCLAMATION, this);
}
}
if (!success) {
SetPlay(false);
SetStop(false);
SetRecord(false);
}
#if (AUDACITY_BRANDING == BRAND_AUDIOTOUCH)
if (success)
{
mPlay->Hide();
mPause->Show();
this->EnablePauseCommand(true);
this->SetBackgroundColour(*wxGREEN); // green for Playing
this->Refresh();
}
#endif
}
}
int main(int argc, char** argv) {
init(argc, argv);
if (argc != 3) {
google::ShowUsageWithFlags(argv[0]);
return 1;
}
std::string tracks_file = argv[1];
std::string image_format = argv[2];
std::string output_format = FLAGS_output_format;
bool ok;
TrackList<DrawerPointer> tracks;
if (FLAGS_similarity) {
// Load tracks.
TrackList<SiftPosition> sift_tracks;
SiftPositionReader feature_reader;
ok = loadTrackList(tracks_file, sift_tracks, feature_reader);
CHECK(ok) << "Could not load tracks";
// Convert SIFT features to generic drawable features.
siftPositionTracksToDrawers(sift_tracks, tracks);
} else if (FLAGS_scale) {
// Load tracks.
TrackList<ScaleSpacePosition> scale_tracks;
ScaleSpacePositionReader feature_reader;
ok = loadTrackList(tracks_file, scale_tracks, feature_reader);
CHECK(ok) << "Could not load tracks";
// Convert SIFT features to generic drawable features.
scaleFeatureTracksToDrawers(scale_tracks, tracks, FLAGS_radius);
} else {
// Load tracks.
TrackList<cv::Point2d> point_tracks;
ImagePointReader<double> feature_reader;
ok = loadTrackList(tracks_file, point_tracks, feature_reader);
CHECK(ok) << "Could not load tracks";
// Convert SIFT features to generic drawable features.
translationTracksToDrawers(point_tracks, tracks, FLAGS_radius);
}
LOG(INFO) << "Loaded " << tracks.size() << " tracks";
// Make a list of random colors.
typedef std::vector<cv::Scalar> ColorList;
ColorList colors;
for (int i = 0; i < int(tracks.size()); i += 1) {
colors.push_back(randomColor(BRIGHTNESS, SATURATION));
}
// Iterate through frames in which track was observed.
TrackListTimeIterator<DrawerPointer> frame(tracks, 0);
while (!frame.end()) {
// Get the current time.
int t = frame.t();
// Load the image.
cv::Mat color_image;
cv::Mat gray_image;
ok = readImage(makeFilename(image_format, t), color_image, gray_image);
CHECK(ok) << "Could not read image";
// Get the features.
typedef std::map<int, DrawerPointer> FeatureSet;
FeatureSet features;
frame.getPoints(features);
// Draw each one with its color.
drawFeatures(color_image, features, colors);
if (FLAGS_save) {
std::string output_file = makeFilename(output_format, t);
ok = cv::imwrite(output_file, color_image);
CHECK(ok) << "Could not save image";
}
if (FLAGS_display) {
cv::imshow("tracks", color_image);
cv::waitKey(10);
}
++frame;
}
return 0;
}
int ExportOGG::Export(AudacityProject *project,
int numChannels,
wxString fName,
bool selectionOnly,
double t0,
double t1,
MixerSpec *mixerSpec,
Tags *metadata,
int WXUNUSED(subformat))
{
double rate = project->GetRate();
TrackList *tracks = project->GetTracks();
double quality = (gPrefs->Read(wxT("/FileFormats/OggExportQuality"), 50)/(float)100.0);
wxLogNull logNo; // temporarily disable wxWidgets error messages
int updateResult = eProgressSuccess;
int eos = 0;
FileIO outFile(fName, FileIO::Output);
if (!outFile.IsOpened()) {
wxMessageBox(_("Unable to open target file for writing"));
return false;
}
// All the Ogg and Vorbis encoding data
ogg_stream_state stream;
ogg_page page;
ogg_packet packet;
vorbis_info info;
vorbis_comment comment;
vorbis_dsp_state dsp;
vorbis_block block;
// Encoding setup
vorbis_info_init(&info);
vorbis_encode_init_vbr(&info, numChannels, int(rate + 0.5), quality);
// Retrieve tags
if (!FillComment(project, &comment, metadata)) {
return false;
}
// Set up analysis state and auxiliary encoding storage
vorbis_analysis_init(&dsp, &info);
vorbis_block_init(&dsp, &block);
// Set up packet->stream encoder. According to encoder example,
// a random serial number makes it more likely that you can make
// chained streams with concatenation.
srand(time(NULL));
ogg_stream_init(&stream, rand());
// First we need to write the required headers:
// 1. The Ogg bitstream header, which contains codec setup params
// 2. The Vorbis comment header
// 3. The bitstream codebook.
//
// After we create those our responsibility is complete, libvorbis will
// take care of any other ogg bistream constraints (again, according
// to the example encoder source)
ogg_packet bitstream_header;
ogg_packet comment_header;
ogg_packet codebook_header;
vorbis_analysis_headerout(&dsp, &comment, &bitstream_header, &comment_header,
&codebook_header);
// Place these headers into the stream
ogg_stream_packetin(&stream, &bitstream_header);
ogg_stream_packetin(&stream, &comment_header);
ogg_stream_packetin(&stream, &codebook_header);
// Flushing these headers now guarentees that audio data will
// start on a new page, which apparently makes streaming easier
while (ogg_stream_flush(&stream, &page)) {
outFile.Write(page.header, page.header_len);
outFile.Write(page.body, page.body_len);
}
int numWaveTracks;
WaveTrack **waveTracks;
tracks->GetWaveTracks(selectionOnly, &numWaveTracks, &waveTracks);
Mixer *mixer = CreateMixer(numWaveTracks, waveTracks,
tracks->GetTimeTrack(),
t0, t1,
numChannels, SAMPLES_PER_RUN, false,
rate, floatSample, true, mixerSpec);
delete [] waveTracks;
ProgressDialog *progress = new ProgressDialog(wxFileName(fName).GetName(),
selectionOnly ?
_("Exporting the selected audio as Ogg Vorbis") :
_("Exporting the entire project as Ogg Vorbis"));
while (updateResult == eProgressSuccess && !eos) {
float **vorbis_buffer = vorbis_analysis_buffer(&dsp, SAMPLES_PER_RUN);
sampleCount samplesThisRun = mixer->Process(SAMPLES_PER_RUN);
if (samplesThisRun == 0) {
// Tell the library that we wrote 0 bytes - signalling the end.
vorbis_analysis_wrote(&dsp, 0);
}
else {
for (int i = 0; i < numChannels; i++) {
float *temp = (float *)mixer->GetBuffer(i);
memcpy(vorbis_buffer[i], temp, sizeof(float)*SAMPLES_PER_RUN);
}
// tell the encoder how many samples we have
vorbis_analysis_wrote(&dsp, samplesThisRun);
}
// I don't understand what this call does, so here is the comment
// from the example, verbatim:
//
// vorbis does some data preanalysis, then divvies up blocks
// for more involved (potentially parallel) processing. Get
// a single block for encoding now
while (vorbis_analysis_blockout(&dsp, &block) == 1) {
// analysis, assume we want to use bitrate management
vorbis_analysis(&block, NULL);
vorbis_bitrate_addblock(&block);
while (vorbis_bitrate_flushpacket(&dsp, &packet)) {
// add the packet to the bitstream
ogg_stream_packetin(&stream, &packet);
// From vorbis-tools-1.0/oggenc/encode.c:
// If we've gone over a page boundary, we can do actual output,
// so do so (for however many pages are available).
while (!eos) {
int result = ogg_stream_pageout(&stream, &page);
if (!result) {
break;
}
outFile.Write(page.header, page.header_len);
outFile.Write(page.body, page.body_len);
if (ogg_page_eos(&page)) {
eos = 1;
}
}
}
}
updateResult = progress->Update(mixer->MixGetCurrentTime()-t0, t1-t0);
}
delete progress;;
delete mixer;
ogg_stream_clear(&stream);
vorbis_block_clear(&block);
vorbis_dsp_clear(&dsp);
vorbis_info_clear(&info);
vorbis_comment_clear(&comment);
outFile.Close();
return updateResult;
}
bool AudacityPrintout::OnPrintPage(int WXUNUSED(page))
{
wxDC *dc = GetDC();
if (!dc)
return false;
int width, height;
dc->GetSize(&width, &height);
int rulerScreenHeight = 40;
int screenTotalHeight = mTracks->GetHeight() + rulerScreenHeight;
double scale = height / (double)screenTotalHeight;
int rulerPageHeight = (int)(rulerScreenHeight * scale);
Ruler ruler;
ruler.SetBounds(0, 0, width, rulerPageHeight);
ruler.SetOrientation(wxHORIZONTAL);
ruler.SetRange(0.0, mTracks->GetEndTime());
ruler.SetFormat(Ruler::TimeFormat);
ruler.SetLabelEdges(true);
ruler.Draw(*dc);
TrackArtist artist;
artist.SetBackgroundBrushes(*wxWHITE_BRUSH, *wxWHITE_BRUSH,
*wxWHITE_PEN, *wxWHITE_PEN);
ViewInfo viewInfo;
viewInfo.selectedRegion = SelectedRegion();
viewInfo.vpos = 0;
viewInfo.h = 0.0;
viewInfo.screen = mTracks->GetEndTime() - viewInfo.h;
viewInfo.total = viewInfo.screen;
viewInfo.zoom = viewInfo.lastZoom = width / viewInfo.screen;
int y = rulerPageHeight;
TrackListIterator iter(mTracks);
Track *n = iter.First();
while (n) {
wxRect r;
r.x = 0;
r.y = y;
r.width = width;
r.height = (int)(n->GetHeight() * scale);
artist.DrawTrack(n, *dc, r, &viewInfo, false, false, false, false);
dc->SetPen(*wxBLACK_PEN);
AColor::Line(*dc, 0, r.y, width, r.y);
#ifdef EXPERIMENTAL_OUTPUT_DISPLAY
if(MONO_WAVE_PAN(n)){
y += r.height;
r.x = 0;
r.y = y;
r.width = width;
r.height = (int)(n->GetHeight(true) * scale);
artist.DrawTrack(n, *dc, r, &viewInfo, false, false, false, false);
dc->SetPen(*wxBLACK_PEN);
AColor::Line(*dc, 0, r.y, width, r.y);
}
#endif
n = iter.Next();
y += r.height;
};
return true;
}
void ControlToolBar::OnRecord(wxCommandEvent &evt)
{
if (gAudioIO->IsBusy()) {
mRecord->PopUp();
return;
}
AudacityProject *p = GetActiveProject();
if (p && p->GetCleanSpeechMode()) {
size_t numProjects = gAudacityProjects.Count();
bool tracks = (p && !p->GetTracks()->IsEmpty());
if (tracks || (numProjects > 1)) {
wxMessageBox(_("Recording in CleanSpeech mode is not possible when a track, or more than one project, is already open."),
_("Recording not permitted"),
wxOK | wxICON_INFORMATION,
this);
mRecord->PopUp();
mRecord->Disable();
return;
}
}
if( evt.GetInt() == 1 ) // used when called by keyboard shortcut. Default (0) ignored.
mRecord->SetShift(true);
if( evt.GetInt() == 2 )
mRecord->SetShift(false);
SetRecord(true);
if (p) {
TrackList *t = p->GetTracks();
TrackListIterator it(t);
if(it.First() == NULL)
mRecord->SetShift(false);
double t0 = p->GetSel0();
double t1 = p->GetSel1();
if (t1 == t0)
t1 = 1000000000.0; // record for a long, long time (tens of years)
/* TODO: set up stereo tracks if that is how the user has set up
* their preferences, and choose sample format based on prefs */
WaveTrackArray newRecordingTracks, playbackTracks;
/* REQUIRES PORTMIDI */
// NoteTrackArray midiTracks;
bool duplex;
gPrefs->Read(wxT("/AudioIO/Duplex"), &duplex, true);
if(duplex){
playbackTracks = t->GetWaveTrackArray(false);
/* REQUIRES PORTMIDI */
// midiTracks = t->GetNoteTrackArray(false);
}
else {
playbackTracks = WaveTrackArray();
/* REQUIRES PORTMIDI */
// midiTracks = NoteTrackArray();
}
// If SHIFT key was down, the user wants append to tracks
int recordingChannels = 0;
bool shifted = mRecord->WasShiftDown();
if (shifted) {
TrackListIterator it(t);
WaveTrack *wt;
bool sel = false;
double allt0 = t0;
// Find the maximum end time of selected and all wave tracks
for (Track *tt = it.First(); tt; tt = it.Next()) {
if (tt->GetKind() == Track::Wave) {
wt = (WaveTrack *)tt;
if (wt->GetEndTime() > allt0) {
allt0 = wt->GetEndTime();
}
if (tt->GetSelected()) {
sel = true;
if (duplex)
playbackTracks.Remove(wt);
if (wt->GetEndTime() > t0) {
t0 = wt->GetEndTime();
}
}
}
}
// Use end time of all wave tracks if none selected
if (!sel) {
t0 = allt0;
}
// Pad selected/all wave tracks to make them all the same length
for (Track *tt = it.First(); tt; tt = it.Next()) {
if (tt->GetKind() == Track::Wave && (tt->GetSelected() || !sel)) {
wt = (WaveTrack *)tt;
t1 = wt->GetEndTime();
if (t1 < t0) {
WaveTrack *newTrack = p->GetTrackFactory()->NewWaveTrack();
newTrack->InsertSilence(0.0, t0 - t1);
newTrack->Flush();
wt->Clear(t1, t0);
wt->Paste(t1, newTrack);
delete newTrack;
}
newRecordingTracks.Add(wt);
}
}
t1 = 1000000000.0; // record for a long, long time (tens of years)
}
else {
recordingChannels = gPrefs->Read(wxT("/AudioIO/RecordChannels"), 2);
for (int c = 0; c < recordingChannels; c++) {
WaveTrack *newTrack = p->GetTrackFactory()->NewWaveTrack();
int initialheight = newTrack->GetHeight();
newTrack->SetOffset(t0);
if (recordingChannels <= 2) {
newTrack->SetHeight(initialheight/recordingChannels);
}
else {
newTrack->SetMinimized(true);
}
if (recordingChannels == 2) {
if (c == 0) {
newTrack->SetChannel(Track::LeftChannel);
newTrack->SetLinked(true);
}
else {
newTrack->SetChannel(Track::RightChannel);
newTrack->SetTeamed(true);
}
}
else {
newTrack->SetChannel( Track::MonoChannel );
}
newRecordingTracks.Add(newTrack);
}
// msmeyer: StartStream calls a callback which triggers auto-save, so
// we add the tracks where recording is done into now. We remove them
// later if starting the stream fails
for (unsigned int i = 0; i < newRecordingTracks.GetCount(); i++)
t->Add(newRecordingTracks[i]);
}
int token = gAudioIO->StartStream(playbackTracks,
newRecordingTracks,
/* REQUIRES PORTMIDI */
// midiTracks,
t->GetTimeTrack(),
p->GetRate(), t0, t1, p);
bool success = (token != 0);
if (success) {
p->SetAudioIOToken(token);
mBusyProject = p;
SetVUMeters(p);
}
else {
// msmeyer: Delete recently added tracks if opening stream fails
if (!shifted) {
for (unsigned int i = 0; i < newRecordingTracks.GetCount(); i++) {
t->Remove(newRecordingTracks[i]);
delete newRecordingTracks[i];
}
}
// msmeyer: Show error message if stream could not be opened
wxMessageBox(_("Error while opening sound device. "
wxT("Please check the input device settings and the project sample rate.")),
_("Error"), wxOK | wxICON_EXCLAMATION, this);
SetPlay(false);
SetStop(false);
SetRecord(false);
}
}
}
void ControlToolBar::PlayPlayRegion(double t0, double t1,
bool looped /* = false */,
bool cutpreview /* = false */,
TimeTrack *timetrack /* = NULL */)
{
SetPlay(true);
if (gAudioIO->IsBusy()) {
SetPlay(false);
return;
}
if (cutpreview && t0==t1) {
SetPlay(false);
return; /* msmeyer: makes no sense */
}
AudacityProject *p = GetActiveProject();
if (!p) {
SetPlay(false);
return; // Should never happen, but...
}
TrackList *t = p->GetTracks();
if (!t) {
mPlay->PopUp();
return; // Should never happen, but...
}
bool hasaudio = false;
TrackListIterator iter(t);
for (Track *trk = iter.First(); trk; trk = iter.Next()) {
if (trk->GetKind() == Track::Wave) {
hasaudio = true;
break;
}
}
if (!hasaudio) {
SetPlay(false);
return; // No need to continue without audio tracks
}
double maxofmins,minofmaxs;
// JS: clarified how the final play region is computed;
if (t1 == t0) {
// msmeyer: When playing looped, we play the whole file, if
// no range is selected. Otherwise, we play from t0 to end
if (looped) {
// msmeyer: always play from start
t0 = t->GetStartTime();
} else {
// move t0 to valid range
if (t0 < 0)
t0 = t->GetStartTime();
if (t0 > t->GetEndTime())
t0 = t->GetEndTime();
}
// always play to end
t1 = t->GetEndTime();
}
else {
// always t0 < t1 right?
// the set intersection between the play region and the
// valid range maximum of lower bounds
if (t0 < t->GetStartTime())
maxofmins = t->GetStartTime();
else
maxofmins = t0;
// minimum of upper bounds
if (t1 > t->GetEndTime())
minofmaxs = t->GetEndTime();
else
minofmaxs = t1;
// we test if the intersection has no volume
if (minofmaxs <= maxofmins) {
// no volume; play nothing
return;
}
else {
t0 = maxofmins;
t1 = minofmaxs;
}
}
// Can't play before 0...either shifted or latencey corrected tracks
if (t0 < 0.0) {
t0 = 0.0;
}
bool success = false;
if (t1 > t0) {
int token;
if (cutpreview) {
double beforeLen, afterLen;
gPrefs->Read(wxT("/AudioIO/CutPreviewBeforeLen"), &beforeLen, 1.0);
gPrefs->Read(wxT("/AudioIO/CutPreviewAfterLen"), &afterLen, 1.0);
double tcp0 = t0-beforeLen;
double tcp1 = (t1+afterLen) - (t1-t0);
SetupCutPreviewTracks(tcp0, t0, t1, tcp1);
if (mCutPreviewTracks)
{
token = gAudioIO->StartStream(
mCutPreviewTracks->GetWaveTrackArray(false),
WaveTrackArray(),
/* REQUIRES PORTMIDI */
// NoteTrackArray(),
NULL, p->GetRate(), tcp0, tcp1, p, false,
t0, t1-t0);
} else
{
// Cannot create cut preview tracks, clean up and exit
SetPlay(false);
SetStop(false);
SetRecord(false);
return;
}
} else {
if (!timetrack) {
timetrack = t->GetTimeTrack();
}
token = gAudioIO->StartStream(t->GetWaveTrackArray(false),
WaveTrackArray(),
/* REQUIRES PORTMIDI */
// t->GetNoteTrackArray(false),
timetrack,
p->GetRate(), t0, t1, p, looped);
}
if (token != 0) {
success = true;
p->SetAudioIOToken(token);
mBusyProject = p;
SetVUMeters(p);
}
else {
// msmeyer: Show error message if stream could not be opened
wxMessageBox(_(
"Error while opening sound device. "
wxT("Please check the output device settings and the project sample rate.")),
_("Error"), wxOK | wxICON_EXCLAMATION, this);
}
}
if (!success) {
SetPlay(false);
SetStop(false);
SetRecord(false);
}
}
