void InsertSilence(COMMAND_T* _ct)
{
char val[64]="";
lstrcpyn(val, g_lastSilenceVal[(int)_ct->user], sizeof(val));
if (PromptUserForString(GetMainHwnd(), SWS_CMD_SHORTNAME(_ct), val, sizeof(val)) && *val)
{
double pos = GetCursorPositionEx(NULL), len = 0.0;
switch ((int)_ct->user)
{
case 0: // s
len = parse_timestr_len(val, pos, 3);
break;
case 1: // meas.beat
{
/* no! would not take get tempo markers into account since 'val' is not inserted yet
len = parse_timestr_len(val, pos, 2);
break;
*/
int in_meas = atoi(val);
double in_beats = 0.0;
char* p = strchr(val, '.');
if (p && p[1]) in_beats = atof(p+1);
double bpm; int num, den;
TimeMap_GetTimeSigAtTime(NULL, pos, &num, &den, &bpm);
len = in_beats*(60.0/bpm) + in_meas*((240.0*num/den)/bpm);
break;
}
case 2: // smp
len = parse_timestr_len(val, pos, 4);
break;
}
if (len>0.0) {
lstrcpyn(g_lastSilenceVal[(int)_ct->user], val, sizeof(val));
InsertSilence(SWS_CMD_SHORTNAME(_ct), pos, len); // includes an undo point
}
else
MessageBox(GetMainHwnd(), __LOCALIZE("Invalid input!","sws_mbox"), __LOCALIZE("S&M - Error","sws_mbox"), MB_OK);
}
}
SmpLength ResizeSample(modplug::tracker::modsample_t& smp, const SmpLength nNewLength, module_renderer* pSndFile)
//--------------------------------------------------------------------------------------
{
// Invalid sample size
if(nNewLength > MAX_SAMPLE_LENGTH || nNewLength == smp.length)
return smp.length;
// New sample will be bigger so we'll just use "InsertSilence" as it's already there.
if(nNewLength > smp.length)
return InsertSilence(smp, nNewLength - smp.length, smp.length, pSndFile);
// Else: Shrink sample
const SmpLength nNewSmpBytes = nNewLength * smp.GetBytesPerSample();
LPSTR pNewSmp = 0;
pNewSmp = module_renderer::AllocateSample(nNewSmpBytes);
if(pNewSmp == 0)
return smp.length; //Sample allocation failed.
// Copy over old data and replace sample by the new one
memcpy(pNewSmp, smp.sample_data, nNewSmpBytes);
ReplaceSample(smp, pNewSmp, nNewLength, pSndFile);
// Adjust loops
if(smp.loop_start > nNewLength)
{
smp.loop_start = smp.loop_end = 0;
smp.flags &= ~CHN_LOOP;
}
if(smp.loop_end > nNewLength) smp.loop_end = nNewLength;
if(smp.sustain_start > nNewLength)
{
smp.sustain_start = smp.sustain_end = 0;
smp.flags &= ~CHN_SUSTAINLOOP;
}
if(smp.sustain_end > nNewLength) smp.sustain_end = nNewLength;
AdjustEndOfSample(smp, pSndFile);
return smp.length;
}
SmpLength ResizeSample(ModSample &smp, const SmpLength nNewLength, CSoundFile &sndFile)
//-------------------------------------------------------------------------------------
{
// Invalid sample size
if(nNewLength > MAX_SAMPLE_LENGTH || nNewLength == smp.nLength)
return smp.nLength;
// New sample will be bigger so we'll just use "InsertSilence" as it's already there.
if(nNewLength > smp.nLength)
return InsertSilence(smp, nNewLength - smp.nLength, smp.nLength, sndFile);
// Else: Shrink sample
const SmpLength nNewSmpBytes = nNewLength * smp.GetBytesPerSample();
void *pNewSmp = ModSample::AllocateSample(nNewLength, smp.GetBytesPerSample());
if(pNewSmp == nullptr)
return smp.nLength; //Sample allocation failed.
// Copy over old data and replace sample by the new one
memcpy(pNewSmp, smp.pSample, nNewSmpBytes);
ReplaceSample(smp, pNewSmp, nNewLength, sndFile);
// Adjust loops
if(smp.nLoopStart > nNewLength)
{
smp.nLoopStart = smp.nLoopEnd = 0;
smp.uFlags.reset(CHN_LOOP);
}
if(smp.nLoopEnd > nNewLength) smp.nLoopEnd = nNewLength;
if(smp.nSustainStart > nNewLength)
{
smp.nSustainStart = smp.nSustainEnd = 0;
smp.uFlags.reset(CHN_SUSTAINLOOP);
}
if(smp.nSustainEnd > nNewLength) smp.nSustainEnd = nNewLength;
PrecomputeLoops(smp, sndFile);
return smp.nLength;
}
/***********************************************************************
* SyncAudio
***********************************************************************
*
**********************************************************************/
static int syncAudioWork( hb_work_object_t * w, hb_buffer_t ** buf_in,
hb_buffer_t ** buf_out )
{
hb_work_private_t * pv = w->private_data;
hb_job_t * job = pv->job;
hb_sync_audio_t * sync = &pv->type.audio;
hb_buffer_t * buf;
int64_t start;
*buf_out = NULL;
buf = *buf_in;
*buf_in = NULL;
/* if the next buffer is an eof send it downstream */
if ( buf->size <= 0 )
{
hb_buffer_close( &buf );
*buf_out = hb_buffer_init( 0 );
pv->common->first_pts[sync->index+1] = INT64_MAX - 1;
return HB_WORK_DONE;
}
/* Wait till we can determine the initial pts of all streams */
if( pv->common->pts_offset == INT64_MIN )
{
pv->common->first_pts[sync->index+1] = buf->s.start;
hb_lock( pv->common->mutex );
while( pv->common->pts_offset == INT64_MIN )
{
// Full fifos will make us wait forever, so get the
// pts offset from the available streams if full
if (hb_fifo_is_full(w->fifo_in))
{
getPtsOffset( w );
hb_cond_broadcast( pv->common->next_frame );
}
else if ( checkPtsOffset( w ) )
hb_cond_broadcast( pv->common->next_frame );
else
hb_cond_timedwait( pv->common->next_frame, pv->common->mutex, 200 );
}
hb_unlock( pv->common->mutex );
}
/* Wait for start frame if doing point-to-point */
hb_lock( pv->common->mutex );
start = buf->s.start - pv->common->audio_pts_slip;
while ( !pv->common->start_found )
{
if ( pv->common->audio_pts_thresh < 0 )
{
// I would initialize this in hb_sync_init, but
// job->pts_to_start can be modified by reader
// after hb_sync_init is called.
pv->common->audio_pts_thresh = job->pts_to_start;
}
if ( buf->s.start < pv->common->audio_pts_thresh )
{
hb_buffer_close( &buf );
hb_unlock( pv->common->mutex );
return HB_WORK_OK;
}
while ( !pv->common->start_found &&
buf->s.start >= pv->common->audio_pts_thresh )
{
hb_cond_timedwait( pv->common->next_frame, pv->common->mutex, 10 );
// There is an unfortunate unavoidable deadlock that can occur.
// Since we need to wait for a specific frame in syncVideoWork,
// syncAudioWork can be stalled indefinitely. The video decoder
// often drops multiple of the initial frames after starting
// because they require references that have not been decoded yet.
// This allows a lot of audio to be queued in the fifo and the
// audio fifo fills before we get a single video frame. So we
// must drop some audio to unplug the pipeline and allow the first
// video frame to be decoded.
if ( hb_fifo_is_full(w->fifo_in) )
{
hb_buffer_t *tmp;
tmp = buf = hb_fifo_get( w->fifo_in );
while ( tmp )
{
tmp = hb_fifo_get( w->fifo_in );
if ( tmp )
{
hb_buffer_close( &buf );
buf = tmp;
}
}
}
}
start = buf->s.start - pv->common->audio_pts_slip;
}
if ( start < 0 )
{
hb_buffer_close( &buf );
hb_unlock( pv->common->mutex );
return HB_WORK_OK;
}
hb_unlock( pv->common->mutex );
if( job->frame_to_stop && pv->common->count_frames >= job->frame_to_stop )
{
hb_buffer_close( &buf );
*buf_out = hb_buffer_init( 0 );
return HB_WORK_DONE;
}
if( job->pts_to_stop && sync->next_start >= job->pts_to_stop )
{
hb_buffer_close( &buf );
*buf_out = hb_buffer_init( 0 );
return HB_WORK_DONE;
}
// audio time went backwards.
// If our output clock is more than a half frame ahead of the
// input clock drop this frame to move closer to sync.
// Otherwise drop frames until the input clock matches the output clock.
if ( sync->next_start - start > 90*15 )
{
// Discard data that's in the past.
if ( sync->first_drop == 0 )
{
sync->first_drop = start;
}
++sync->drop_count;
hb_buffer_close( &buf );
return HB_WORK_OK;
}
if ( sync->first_drop )
{
// we were dropping old data but input buf time is now current
hb_log( "sync: audio 0x%x time went backwards %d ms, dropped %d frames "
"(start %"PRId64", next %"PRId64")", w->audio->id,
(int)( sync->next_start - sync->first_drop ) / 90,
sync->drop_count, sync->first_drop, (int64_t)sync->next_start );
sync->first_drop = 0;
sync->drop_count = 0;
}
if ( start - sync->next_start >= (90 * 70) )
{
if ( start - sync->next_start > (90000LL * 60) )
{
// there's a gap of more than a minute between the last
// frame and this. assume we got a corrupted timestamp
// and just drop the next buf.
hb_log( "sync: %d minute time gap in audio 0x%x - dropping buf"
" start %"PRId64", next %"PRId64,
(int)((start - sync->next_start) / (90000*60)),
w->audio->id, start, (int64_t)sync->next_start );
hb_buffer_close( &buf );
return HB_WORK_OK;
}
/*
* there's a gap of at least 70ms between the last
* frame we processed & the next. Fill it with silence.
* Or in the case of DCA, skip some frames from the
* other streams.
*/
if ( sync->drop_video_to_sync )
{
hb_log( "sync: audio gap %d ms. Skipping frames. Audio 0x%x"
" start %"PRId64", next %"PRId64,
(int)((start - sync->next_start) / 90),
w->audio->id, start, (int64_t)sync->next_start );
hb_lock( pv->common->mutex );
pv->common->audio_pts_slip += (start - sync->next_start);
pv->common->video_pts_slip += (start - sync->next_start);
hb_unlock( pv->common->mutex );
*buf_out = OutputAudioFrame( w->audio, buf, sync );
return HB_WORK_OK;
}
hb_log( "sync: adding %d ms of silence to audio 0x%x"
" start %"PRId64", next %"PRId64,
(int)((start - sync->next_start) / 90),
w->audio->id, start, (int64_t)sync->next_start );
InsertSilence( w, start - sync->next_start );
}
/*
* When we get here we've taken care of all the dups and gaps in the
* audio stream and are ready to inject the next input frame into
* the output stream.
*/
*buf_out = OutputAudioFrame( w->audio, buf, sync );
return HB_WORK_OK;
}
void ControlToolBar::OnRecord(wxCommandEvent &evt)
{
auto doubleClicked = mRecord->IsDoubleClicked();
mRecord->ClearDoubleClicked();
if (doubleClicked) {
// Display a fixed recording head while scrolling the waves continuously.
// If you overdub, you may want to anticipate some context in existing tracks,
// so center the head. If not, put it rightmost to display as much wave as we can.
const auto project = GetActiveProject();
bool duplex;
gPrefs->Read(wxT("/AudioIO/Duplex"), &duplex, true);
if (duplex) {
// See if there is really anything being overdubbed
if (gAudioIO->GetNumPlaybackChannels() == 0)
// No.
duplex = false;
}
using Mode = AudacityProject::PlaybackScroller::Mode;
project->GetPlaybackScroller().Activate(duplex ? Mode::Centered : Mode::Right);
return;
}
if (gAudioIO->IsBusy()) {
if (!CanStopAudioStream() || 0 == gAudioIO->GetNumCaptureChannels())
mRecord->PopUp();
else
mRecord->PushDown();
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 *trackList = p->GetTracks();
TrackListIterator it(trackList);
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 = trackList->GetWaveTrackArray(false);
#ifdef EXPERIMENTAL_MIDI_OUT
midiTracks = trackList->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;
TrackList tracksCopy{};
bool tracksCopied = false;
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) {
auto end = playbackTracks.end();
auto it = std::find(playbackTracks.begin(), end, wt);
if (it != end)
playbackTracks.erase(it);
}
t1 = wt->GetEndTime();
if (t1 < t0) {
if (!tracksCopied) {
tracksCopied = true;
tracksCopy = *trackList;
}
auto newTrack = p->GetTrackFactory()->NewWaveTrack();
newTrack->InsertSilence(0.0, t0 - t1);
newTrack->Flush();
wt->Clear(t1, t0);
bool bResult = wt->Paste(t1, newTrack.get());
wxASSERT(bResult); // TO DO: Actually handle this.
wxUnusedVar(bResult);
}
newRecordingTracks.push_back(wt);
}
}
t1 = 1000000000.0; // record for a long, long time (tens of years)
}
else {
bool recordingNameCustom, useTrackNumber, useDateStamp, useTimeStamp;
wxString defaultTrackName, defaultRecordingTrackName;
int numTracks = 0;
for (Track *tt = it.First(); tt; tt = it.Next()) {
if (tt->GetKind() == Track::Wave && !tt->GetLinked())
numTracks++;
}
numTracks++;
recordingChannels = gPrefs->Read(wxT("/AudioIO/RecordChannels"), 2);
gPrefs->Read(wxT("/GUI/TrackNames/RecordingNameCustom"), &recordingNameCustom, false);
gPrefs->Read(wxT("/GUI/TrackNames/TrackNumber"), &useTrackNumber, false);
gPrefs->Read(wxT("/GUI/TrackNames/DateStamp"), &useDateStamp, false);
gPrefs->Read(wxT("/GUI/TrackNames/TimeStamp"), &useTimeStamp, false);
/* i18n-hint: The default name for an audio track. */
gPrefs->Read(wxT("/GUI/TrackNames/DefaultTrackName"),&defaultTrackName, _("Audio Track"));
gPrefs->Read(wxT("/GUI/TrackNames/RecodingTrackName"), &defaultRecordingTrackName, defaultTrackName);
wxString baseTrackName = recordingNameCustom? defaultRecordingTrackName : defaultTrackName;
for (int c = 0; c < recordingChannels; c++) {
auto newTrack = p->GetTrackFactory()->NewWaveTrack();
newTrack->SetOffset(t0);
wxString nameSuffix = wxString(wxT(""));
if (useTrackNumber) {
nameSuffix += wxString::Format(wxT("%d"), numTracks + c);
}
if (useDateStamp) {
if (!nameSuffix.IsEmpty()) {
nameSuffix += wxT("_");
}
nameSuffix += wxDateTime::Now().FormatISODate();
}
if (useTimeStamp) {
if (!nameSuffix.IsEmpty()) {
nameSuffix += wxT("_");
}
nameSuffix += wxDateTime::Now().FormatISOTime();
}
// ISO standard would be nice, but ":" is unsafe for file name.
nameSuffix.Replace(wxT(":"), wxT("-"));
if (baseTrackName.IsEmpty()) {
newTrack->SetName(nameSuffix);
}
else if (nameSuffix.IsEmpty()) {
newTrack->SetName(baseTrackName);
}
else {
newTrack->SetName(baseTrackName + wxT("_") + nameSuffix);
}
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 );
}
// Let the list hold the track, and keep a pointer to it
newRecordingTracks.push_back(
static_cast<WaveTrack*>(
trackList->Add(
std::move(newTrack))));
}
}
//Automated Input Level Adjustment Initialization
#ifdef EXPERIMENTAL_AUTOMATED_INPUT_LEVEL_ADJUSTMENT
gAudioIO->AILAInitialize();
#endif
AudioIOStartStreamOptions options(p->GetDefaultPlayOptions());
int token = gAudioIO->StartStream(playbackTracks,
newRecordingTracks,
#ifdef EXPERIMENTAL_MIDI_OUT
midiTracks,
#endif
t0, t1, options);
bool success = (token != 0);
if (success) {
p->SetAudioIOToken(token);
mBusyProject = p;
}
else {
if (shifted) {
// Restore the tracks to remove any inserted silence
if (tracksCopied)
*trackList = std::move(tracksCopy);
}
else {
// msmeyer: Delete recently added tracks if opening stream fails
for (unsigned int i = 0; i < newRecordingTracks.size(); i++) {
trackList->Remove(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);
}
}
UpdateStatusBar(GetActiveProject());
}
