bool EffectSilence::Process()
{
if (length <= 0.0)
length = sDefaultGenerateLen;
TrackListIterator iter(mWaveTracks);
WaveTrack *track = (WaveTrack *)iter.First();
while (track) {
WaveTrack *tmp = mFactory->NewWaveTrack(track->GetSampleFormat(), track->GetRate());
tmp->InsertSilence(0.0, length);
tmp->Flush();
track->Clear(mT0, mT1);
track->Paste(mT0, tmp);
delete tmp;
//Iterate to the next track
track = (WaveTrack *)iter.Next();
}
mT1 = mT0 + length; // Update selection.
return true;
}
int FFmpegImportFileHandle::Import(TrackFactory *trackFactory,
Track ***outTracks,
int *outNumTracks,
Tags *tags)
{
CreateProgress();
// Remove stream contexts which are not marked for importing and adjust mScs and mNumStreams accordingly
for (int i = 0; i < mNumStreams;)
{
if (!mScs[i]->m_use)
{
delete mScs[i];
for (int j = i; j < mNumStreams - 1; j++)
{
mScs[j] = mScs[j+1];
}
mNumStreams--;
}
else i++;
}
mChannels = new WaveTrack **[mNumStreams];
mNumSamples = 0;
for (int s = 0; s < mNumStreams; s++)
{
// As you can see, it's really a number of frames.
// TODO: use something other than nb_frames for progress reporting (nb_frames is not available for some formats). Maybe something from the format context?
mNumSamples += mScs[s]->m_stream->nb_frames;
// There is a possibility that number of channels will change over time, but we do not have WaveTracks for new channels. Remember the number of channels and stick to it.
mScs[s]->m_initialchannels = mScs[s]->m_stream->codec->channels;
mChannels[s] = new WaveTrack *[mScs[s]->m_stream->codec->channels];
int c;
for (c = 0; c < mScs[s]->m_stream->codec->channels; c++)
{
mChannels[s][c] = trackFactory->NewWaveTrack(int16Sample, mScs[s]->m_stream->codec->sample_rate);
if (mScs[s]->m_stream->codec->channels == 2)
{
switch (c)
{
case 0:
mChannels[s][c]->SetChannel(Track::LeftChannel);
mChannels[s][c]->SetLinked(true);
break;
case 1:
mChannels[s][c]->SetChannel(Track::RightChannel);
mChannels[s][c]->SetTeamed(true);
break;
}
}
else
{
mChannels[s][c]->SetChannel(Track::MonoChannel);
}
}
}
// Handles the start_time by creating silence. This may or may not be correct.
// There is a possibility that we should ignore first N milliseconds of audio instead. I do not know.
/// TODO: Nag FFmpeg devs about start_time until they finally say WHAT is this and HOW to handle it.
for (int s = 0; s < mNumStreams; s++)
{
int64_t stream_delay = 0;
if (mScs[s]->m_stream->start_time != int64_t(AV_NOPTS_VALUE))
{
stream_delay = mScs[s]->m_stream->start_time;
wxLogMessage(wxT("Stream %d start_time = %d, that would be %f milliseconds."), s, mScs[s]->m_stream->start_time, double(mScs[s]->m_stream->start_time)/AV_TIME_BASE*1000);
}
if (stream_delay != 0)
{
for (int c = 0; c < mScs[s]->m_stream->codec->channels; c++)
{
WaveTrack *t = mChannels[s][c];
t->InsertSilence(0,double(stream_delay)/AV_TIME_BASE);
}
}
}
// This is the heart of the importing process
streamContext *sc = NULL;
// The result of Import() to be returend. It will be something other than zero if user canceled or some error appears.
int res = 0;
// Read next frame.
while ((sc = ReadNextFrame()) != NULL && (res == 0))
{
// ReadNextFrame returns 1 if stream is not to be imported
if (sc != (streamContext*)1)
{
// Decode frame until it is not possible to decode any further
while (sc->m_pktRemainingSiz > 0 && (res == 0))
{
if (DecodeFrame(sc,false) < 0)
break;
// If something useable was decoded - write it to mChannels
if (sc->m_frameValid)
res = WriteData(sc);
}
// Cleanup after frame decoding
if (sc->m_pktValid)
{
av_free_packet(&sc->m_pkt);
sc->m_pktValid = 0;
}
}
}
// Flush the decoders.
if ((mNumStreams != 0) && (res == 0))
{
for (int i = 0; i < mNumStreams; i++)
{
if (DecodeFrame(mScs[i], true) == 0)
{
WriteData(mScs[i]);
if (mScs[i]->m_pktValid)
{
av_free_packet(&mScs[i]->m_pkt);
mScs[i]->m_pktValid = 0;
}
}
}
}
// Something bad happened - destroy everything!
if (res)
{
for (int s = 0; s < mNumStreams; s++)
{
delete[] mChannels[s];
}
delete[] mChannels;
if (mCancelled)
return eImportCancelled;
else
return eImportFailed;
}
*outNumTracks = 0;
for (int s = 0; s < mNumStreams; s++)
{
*outNumTracks += mScs[s]->m_stream->codec->channels;
}
// Create new tracks
*outTracks = new Track *[*outNumTracks];
// Copy audio from mChannels to newly created tracks (destroying mChannels elements in process)
int trackindex = 0;
for (int s = 0; s < mNumStreams; s++)
{
for(int c = 0; c < mScs[s]->m_stream->codec->channels; c++)
{
mChannels[s][c]->Flush();
(*outTracks)[trackindex++] = mChannels[s][c];
}
delete[] mChannels[s];
}
delete[] mChannels;
// Save metadata
WriteMetadata(mFormatContext,tags);
return eImportSuccess;
}
int FFmpegImportFileHandle::Import(TrackFactory *trackFactory,
Track ***outTracks,
int *outNumTracks,
Tags *tags)
{
CreateProgress();
// Remove stream contexts which are not marked for importing and adjust mScs and mNumStreams accordingly
for (int i = 0; i < mNumStreams;)
{
if (!mScs[i]->m_use)
{
delete mScs[i];
for (int j = i; j < mNumStreams - 1; j++)
{
mScs[j] = mScs[j+1];
}
mNumStreams--;
}
else i++;
}
mChannels = new WaveTrack **[mNumStreams];
for (int s = 0; s < mNumStreams; s++)
{
switch (mScs[s]->m_stream->codec->sample_fmt)
{
case SAMPLE_FMT_U8:
case SAMPLE_FMT_S16:
mScs[s]->m_osamplesize = sizeof(int16_t);
mScs[s]->m_osamplefmt = int16Sample;
break;
default:
mScs[s]->m_osamplesize = sizeof(float);
mScs[s]->m_osamplefmt = floatSample;
break;
}
// There is a possibility that number of channels will change over time, but we do not have WaveTracks for new channels. Remember the number of channels and stick to it.
mScs[s]->m_initialchannels = mScs[s]->m_stream->codec->channels;
mChannels[s] = new WaveTrack *[mScs[s]->m_stream->codec->channels];
int c;
for (c = 0; c < mScs[s]->m_stream->codec->channels; c++)
{
mChannels[s][c] = trackFactory->NewWaveTrack(mScs[s]->m_osamplefmt, mScs[s]->m_stream->codec->sample_rate);
if (mScs[s]->m_stream->codec->channels == 2)
{
switch (c)
{
case 0:
mChannels[s][c]->SetChannel(Track::LeftChannel);
mChannels[s][c]->SetLinked(true);
break;
case 1:
mChannels[s][c]->SetChannel(Track::RightChannel);
break;
}
}
else
{
mChannels[s][c]->SetChannel(Track::MonoChannel);
}
}
}
// Handles the start_time by creating silence. This may or may not be correct.
// There is a possibility that we should ignore first N milliseconds of audio instead. I do not know.
/// TODO: Nag FFmpeg devs about start_time until they finally say WHAT is this and HOW to handle it.
for (int s = 0; s < mNumStreams; s++)
{
int64_t stream_delay = 0;
if (mScs[s]->m_stream->start_time != int64_t(AV_NOPTS_VALUE) && mScs[s]->m_stream->start_time > 0)
{
stream_delay = mScs[s]->m_stream->start_time;
wxLogDebug(wxT("Stream %d start_time = %d, that would be %f milliseconds."), s, mScs[s]->m_stream->start_time, double(mScs[s]->m_stream->start_time)/AV_TIME_BASE*1000);
}
if (stream_delay != 0)
{
for (int c = 0; c < mScs[s]->m_stream->codec->channels; c++)
{
WaveTrack *t = mChannels[s][c];
t->InsertSilence(0,double(stream_delay)/AV_TIME_BASE);
}
}
}
// This is the heart of the importing process
// The result of Import() to be returend. It will be something other than zero if user canceled or some error appears.
int res = eProgressSuccess;
#ifdef EXPERIMENTAL_OD_FFMPEG
mUsingOD = false;
gPrefs->Read(wxT("/Library/FFmpegOnDemand"), &mUsingOD);
//at this point we know the file is good and that we have to load the number of channels in mScs[s]->m_stream->codec->channels;
//so for OD loading we create the tracks and releasee the modal lock after starting the ODTask.
if (mUsingOD) {
std::vector<ODDecodeFFmpegTask*> tasks;
//append blockfiles to each stream and add an individual ODDecodeTask for each one.
for (int s = 0; s < mNumStreams; s++) {
ODDecodeFFmpegTask* odTask=new ODDecodeFFmpegTask(mScs,mNumStreams,mChannels,mFormatContext, s);
odTask->CreateFileDecoder(mFilename);
//each stream has different duration. We need to know it if seeking is to be allowed.
sampleCount sampleDuration = 0;
if (mScs[s]->m_stream->duration > 0)
sampleDuration = ((sampleCount)mScs[s]->m_stream->duration * mScs[s]->m_stream->time_base.num) *mScs[s]->m_stream->codec->sample_rate / mScs[s]->m_stream->time_base.den;
else
sampleDuration = ((sampleCount)mFormatContext->duration *mScs[s]->m_stream->codec->sample_rate) / AV_TIME_BASE;
// printf(" OD duration samples %qi, sr %d, secs %d\n",sampleDuration, (int)mScs[s]->m_stream->codec->sample_rate,(int)sampleDuration/mScs[s]->m_stream->codec->sample_rate);
//for each wavetrack within the stream add coded blockfiles
for (int c = 0; c < mScs[s]->m_stream->codec->channels; c++) {
WaveTrack *t = mChannels[s][c];
odTask->AddWaveTrack(t);
sampleCount maxBlockSize = t->GetMaxBlockSize();
//use the maximum blockfile size to divide the sections (about 11secs per blockfile at 44.1khz)
for (sampleCount i = 0; i < sampleDuration; i += maxBlockSize) {
sampleCount blockLen = maxBlockSize;
if (i + blockLen > sampleDuration)
blockLen = sampleDuration - i;
t->AppendCoded(mFilename, i, blockLen, c,ODTask::eODFFMPEG);
// This only works well for single streams since we assume
// each stream is of the same duration and channels
res = mProgress->Update(i+sampleDuration*c+ sampleDuration*mScs[s]->m_stream->codec->channels*s,
sampleDuration*mScs[s]->m_stream->codec->channels*mNumStreams);
if (res != eProgressSuccess)
break;
}
}
tasks.push_back(odTask);
}
//Now we add the tasks and let them run, or delete them if the user cancelled
for(int i=0; i < (int)tasks.size(); i++) {
if(res==eProgressSuccess)
ODManager::Instance()->AddNewTask(tasks[i]);
else
{
delete tasks[i];
}
}
} else {
#endif
streamContext *sc = NULL;
// Read next frame.
while ((sc = ReadNextFrame()) != NULL && (res == eProgressSuccess))
{
// ReadNextFrame returns 1 if stream is not to be imported
if (sc != (streamContext*)1)
{
// Decode frame until it is not possible to decode any further
while (sc->m_pktRemainingSiz > 0 && (res == eProgressSuccess || res == eProgressStopped))
{
if (DecodeFrame(sc,false) < 0)
break;
// If something useable was decoded - write it to mChannels
if (sc->m_frameValid)
res = WriteData(sc);
}
// Cleanup after frame decoding
if (sc->m_pktValid)
{
av_free_packet(&sc->m_pkt);
sc->m_pktValid = 0;
}
}
}
// Flush the decoders.
if ((mNumStreams != 0) && (res == eProgressSuccess || res == eProgressStopped))
{
for (int i = 0; i < mNumStreams; i++)
{
if (DecodeFrame(mScs[i], true) == 0)
{
WriteData(mScs[i]);
if (mScs[i]->m_pktValid)
{
av_free_packet(&mScs[i]->m_pkt);
mScs[i]->m_pktValid = 0;
}
}
}
}
#ifdef EXPERIMENTAL_OD_FFMPEG
} // else -- !mUsingOD == true
#endif //EXPERIMENTAL_OD_FFMPEG
// Something bad happened - destroy everything!
if (res == eProgressCancelled || res == eProgressFailed)
{
for (int s = 0; s < mNumStreams; s++)
{
delete[] mChannels[s];
}
delete[] mChannels;
return res;
}
//else if (res == 2), we just stop the decoding as if the file has ended
*outNumTracks = 0;
for (int s = 0; s < mNumStreams; s++)
{
*outNumTracks += mScs[s]->m_initialchannels;
}
// Create new tracks
*outTracks = new Track *[*outNumTracks];
// Copy audio from mChannels to newly created tracks (destroying mChannels elements in process)
int trackindex = 0;
for (int s = 0; s < mNumStreams; s++)
{
for(int c = 0; c < mScs[s]->m_initialchannels; c++)
{
mChannels[s][c]->Flush();
(*outTracks)[trackindex++] = mChannels[s][c];
}
delete[] mChannels[s];
}
delete[] mChannels;
// Save metadata
WriteMetadata(tags);
return res;
}
void ControlToolBar::OnRecord(wxCommandEvent &evt)
{
if (gAudioIO->IsBusy()) {
mRecord->PopUp();
return;
}
AudacityProject *p = GetActiveProject();
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, mRecord->WasShiftDown());
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;
#ifdef EXPERIMENTAL_MIDI_OUT
NoteTrackArray midiTracks;
#endif
bool duplex;
gPrefs->Read(wxT("/AudioIO/Duplex"), &duplex, true);
if(duplex){
playbackTracks = t->GetWaveTrackArray(false);
#ifdef EXPERIMENTAL_MIDI_OUT
midiTracks = t->GetNoteTrackArray(false);
#endif
}
else {
playbackTracks = WaveTrackArray();
#ifdef EXPERIMENTAL_MIDI_OUT
midiTracks = NoteTrackArray();
#endif
}
// If SHIFT key was down, the user wants append to tracks
int recordingChannels = 0;
bool shifted = mRecord->WasShiftDown();
if (shifted) {
bool sel = false;
double allt0 = t0;
// Find the maximum end time of selected and all wave tracks
// Find whether any tracks were selected. (If any are selected,
// record only into them; else if tracks exist, record into all.)
for (Track *tt = it.First(); tt; tt = it.Next()) {
if (tt->GetKind() == Track::Wave) {
WaveTrack *wt = static_cast<WaveTrack *>(tt);
if (wt->GetEndTime() > allt0) {
allt0 = wt->GetEndTime();
}
if (tt->GetSelected()) {
sel = true;
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
// Remove recording tracks from the list of tracks for duplex ("overdub")
// playback.
for (Track *tt = it.First(); tt; tt = it.Next()) {
if (tt->GetKind() == Track::Wave && (tt->GetSelected() || !sel)) {
WaveTrack *wt = static_cast<WaveTrack *>(tt);
if (duplex)
playbackTracks.Remove(wt);
t1 = wt->GetEndTime();
if (t1 < t0) {
WaveTrack *newTrack = p->GetTrackFactory()->NewWaveTrack();
newTrack->InsertSilence(0.0, t0 - t1);
newTrack->Flush();
wt->Clear(t1, t0);
bool bResult = wt->Paste(t1, newTrack);
wxASSERT(bResult); // TO DO: Actually handle this.
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();
newTrack->SetOffset(t0);
if (recordingChannels > 2)
newTrack->SetMinimized(true);
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);
}
// 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]);
}
//Automated Input Level Adjustment Initialization
#ifdef AUTOMATED_INPUT_LEVEL_ADJUSTMENT
gAudioIO->AILAInitialize();
#endif
int token = gAudioIO->StartStream(playbackTracks,
newRecordingTracks,
#ifdef EXPERIMENTAL_MIDI_OUT
midiTracks,
#endif
t->GetTimeTrack(),
p->GetRate(), t0, t1, p);
bool success = (token != 0);
if (success) {
p->SetAudioIOToken(token);
mBusyProject = 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. Please check the recording device settings and the project sample rate."),
_("Error"), wxOK | wxICON_EXCLAMATION, this);
SetPlay(false);
SetStop(false);
SetRecord(false);
}
}
}
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);
}
}
}