bool MixAndRender(TrackList *tracks, TrackFactory *trackFactory,
double rate, sampleFormat format,
double startTime, double endTime,
WaveTrack **newLeft, WaveTrack **newRight)
{
// This function was formerly known as "Quick Mix". It takes one or
// more tracks as input; of all tracks that are selected, it mixes
// them together, applying any envelopes, amplitude gain, panning,
// and real-time effects in the process. The resulting pair of
// tracks (stereo) are "rendered" and have no effects, gain, panning,
// or envelopes.
WaveTrack **waveArray;
Track *t;
int numWaves = 0;
int numMono = 0;
bool mono = false;
int w;
TrackListIterator iter(tracks);
t = iter.First();
while (t) {
if (t->GetSelected() && t->GetKind() == Track::Wave) {
numWaves++;
float pan = ((WaveTrack*)t)->GetPan();
if (t->GetChannel() == Track::MonoChannel && pan == 0)
numMono++;
}
t = iter.Next();
}
if (numMono == numWaves)
mono = true;
double totalTime = 0.0;
waveArray = new WaveTrack *[numWaves];
w = 0;
t = iter.First();
while (t) {
if (t->GetSelected() && t->GetKind() == Track::Wave) {
waveArray[w++] = (WaveTrack *) t;
if (t->GetEndTime() > totalTime)
totalTime = t->GetEndTime();
}
t = iter.Next();
}
WaveTrack *mixLeft = trackFactory->NewWaveTrack(format, rate);
mixLeft->SetName(_("Mix"));
WaveTrack *mixRight = 0;
if (mono) {
mixLeft->SetChannel(Track::MonoChannel);
}
else {
mixRight = trackFactory->NewWaveTrack(format, rate);
mixRight->SetName(_("Mix"));
mixLeft->SetChannel(Track::LeftChannel);
mixRight->SetChannel(Track::RightChannel);
mixLeft->SetLinked(true);
mixRight->SetTeamed(true);
}
int maxBlockLen = mixLeft->GetIdealBlockSize();
if (startTime == endTime) {
startTime = 0.0;
endTime = totalTime;
}
Mixer *mixer = new Mixer(numWaves, waveArray, tracks->GetTimeTrack(),
startTime, endTime, mono ? 1 : 2, maxBlockLen, false,
rate, format);
wxYield();
GetActiveProject()->ProgressShow(_NoAcc("&Mix and Render"),
_("Mixing and rendering tracks"));
wxBusyCursor busy;
bool cancelling = false;
while(!cancelling) {
sampleCount blockLen = mixer->Process(maxBlockLen);
if (blockLen == 0)
break;
if (mono) {
samplePtr buffer = mixer->GetBuffer();
mixLeft->Append(buffer, format, blockLen);
}
else {
samplePtr buffer;
buffer = mixer->GetBuffer(0);
mixLeft->Append(buffer, format, blockLen);
buffer = mixer->GetBuffer(1);
mixRight->Append(buffer, format, blockLen);
}
int progressvalue = int (1000 * (mixer->MixGetCurrentTime() / totalTime));
cancelling = !GetActiveProject()->ProgressUpdate(progressvalue);
}
GetActiveProject()->ProgressHide();
mixLeft->Flush();
*newLeft = mixLeft;
if (!mono) {
mixRight->Flush();
*newRight = mixRight;
}
#if 0
int elapsedMS = wxGetElapsedTime();
double elapsedTime = elapsedMS * 0.001;
double maxTracks = totalTime / (elapsedTime / numWaves);
// Note: these shouldn't be translated - they're for debugging
// and profiling only.
printf(" Tracks: %d\n", numWaves);
printf(" Mix length: %f sec\n", totalTime);
printf("Elapsed time: %f sec\n", elapsedTime);
printf("Max number of tracks to mix in real time: %f\n", maxTracks);
#endif
delete[] waveArray;
delete mixer;
return true;
}
int ExportFLAC::Export(AudacityProject *project,
int numChannels,
const wxString &fName,
bool selectionOnly,
double t0,
double t1,
MixerSpec *mixerSpec,
const Tags *metadata,
int WXUNUSED(subformat))
{
double rate = project->GetRate();
TrackList *tracks = project->GetTracks();
wxLogNull logNo; // temporarily disable wxWidgets error messages
int updateResult = eProgressSuccess;
int levelPref;
gPrefs->Read(wxT("/FileFormats/FLACLevel"), &levelPref, 5);
wxString bitDepthPref =
gPrefs->Read(wxT("/FileFormats/FLACBitDepth"), wxT("16"));
FLAC::Encoder::File encoder;
#ifdef LEGACY_FLAC
encoder.set_filename(OSOUTPUT(fName));
#endif
encoder.set_channels(numChannels);
encoder.set_sample_rate(lrint(rate));
// See note in GetMetadata() about a bug in libflac++ 1.1.2
if (!GetMetadata(project, metadata)) {
return false;
}
if (mMetadata) {
encoder.set_metadata(&mMetadata, 1);
}
sampleFormat format;
if (bitDepthPref == wxT("24")) {
format = int24Sample;
encoder.set_bits_per_sample(24);
} else { //convert float to 16 bits
format = int16Sample;
encoder.set_bits_per_sample(16);
}
// Duplicate the flac command line compression levels
if (levelPref < 0 || levelPref > 8) {
levelPref = 5;
}
encoder.set_do_exhaustive_model_search(flacLevels[levelPref].do_exhaustive_model_search);
encoder.set_do_escape_coding(flacLevels[levelPref].do_escape_coding);
if (numChannels != 2) {
encoder.set_do_mid_side_stereo(false);
encoder.set_loose_mid_side_stereo(false);
}
else {
encoder.set_do_mid_side_stereo(flacLevels[levelPref].do_mid_side_stereo);
encoder.set_loose_mid_side_stereo(flacLevels[levelPref].loose_mid_side_stereo);
}
encoder.set_qlp_coeff_precision(flacLevels[levelPref].qlp_coeff_precision);
encoder.set_min_residual_partition_order(flacLevels[levelPref].min_residual_partition_order);
encoder.set_max_residual_partition_order(flacLevels[levelPref].max_residual_partition_order);
encoder.set_rice_parameter_search_dist(flacLevels[levelPref].rice_parameter_search_dist);
encoder.set_max_lpc_order(flacLevels[levelPref].max_lpc_order);
#ifdef LEGACY_FLAC
encoder.init();
#else
wxFFile f; // will be closed when it goes out of scope
if (!f.Open(fName, wxT("w+b"))) {
wxMessageBox(wxString::Format(_("FLAC export couldn't open %s"), fName.c_str()));
return false;
}
// Even though there is an init() method that takes a filename, use the one that
// takes a file handle because wxWidgets can open a file with a Unicode name and
// libflac can't (under Windows).
int status = encoder.init(f.fp());
if (status != FLAC__STREAM_ENCODER_INIT_STATUS_OK) {
wxMessageBox(wxString::Format(_("FLAC encoder failed to initialize\nStatus: %d"), status));
return false;
}
#endif
if (mMetadata) {
::FLAC__metadata_object_delete(mMetadata);
}
int numWaveTracks;
WaveTrack **waveTracks;
tracks->GetWaveTracks(selectionOnly, &numWaveTracks, &waveTracks);
Mixer *mixer = CreateMixer(numWaveTracks, waveTracks,
tracks->GetTimeTrack(),
t0, t1,
numChannels, SAMPLES_PER_RUN, false,
rate, format, true, mixerSpec);
delete [] waveTracks;
int i, j;
FLAC__int32 **tmpsmplbuf = new FLAC__int32*[numChannels];
for (i = 0; i < numChannels; i++) {
tmpsmplbuf[i] = (FLAC__int32 *) calloc(SAMPLES_PER_RUN, sizeof(FLAC__int32));
}
{
ProgressDialog progress(wxFileName(fName).GetName(),
selectionOnly ?
_("Exporting the selected audio as FLAC") :
_("Exporting the entire project as FLAC"));
while (updateResult == eProgressSuccess) {
sampleCount samplesThisRun = mixer->Process(SAMPLES_PER_RUN);
if (samplesThisRun == 0) { //stop encoding
break;
}
else {
for (i = 0; i < numChannels; i++) {
samplePtr mixed = mixer->GetBuffer(i);
if (format == int24Sample) {
for (j = 0; j < samplesThisRun; j++) {
tmpsmplbuf[i][j] = ((int *)mixed)[j];
}
}
else {
for (j = 0; j < samplesThisRun; j++) {
tmpsmplbuf[i][j] = ((short *)mixed)[j];
}
}
}
encoder.process(tmpsmplbuf, samplesThisRun);
}
updateResult = progress.Update(mixer->MixGetCurrentTime() - t0, t1 - t0);
}
f.Detach(); // libflac closes the file
encoder.finish();
}
for (i = 0; i < numChannels; i++) {
free(tmpsmplbuf[i]);
}
delete mixer;
delete[] tmpsmplbuf;
return updateResult;
}
bool PCMExporter::Export(const wxString & filename)
{
mFileName = filename;
//Unless it has been changed, use the project rate
if(!mOutRate)
mOutRate = mProject->GetRate();
wxWindow *parent = mProject;
TrackList *tracks = mProject->GetTracks();
wxString formatStr;
SF_INFO info;
SNDFILE *sf = NULL;
int err;
formatStr = sf_header_name(mSoundFileFormat & SF_FORMAT_TYPEMASK);
// Use libsndfile to export file
info.samplerate = mOutRate;
info.frames = (unsigned int)((m_t1 - m_t0)*(double)mOutRate + 0.5);
info.channels = mChannels;
info.format = mSoundFileFormat;
info.sections = 1;
info.seekable = 0;
// If we can't export exactly the format they requested,
// try the default format for that header type...
if (!sf_format_check(&info))
info.format = (info.format & SF_FORMAT_TYPEMASK);
if (!sf_format_check(&info)) {
wxMessageBox(_("Cannot export audio in this format."));
return false;
}
sf = sf_open((const char *)mFileName, SFM_WRITE, &info);
if (!sf) {
wxMessageBox(wxString::Format(_("Cannot export audio to %s"),
(const char *)mFileName));
return false;
}
//Determine what format the track is currently in, to avoid
//multiple conversions.
sampleFormat format;
if (sf_subtype_more_than_16_bits(info.format))
format = floatSample;
else
format = int16Sample;
int maxBlockLen = 44100 * 5;
wxProgressDialog *progress = NULL;
wxYield();
wxStartTimer();
wxBusyCursor busy;
bool cancelling = false;
int numWaveTracks;
WaveTrack **waveTracks;
tracks->GetWaveTracks(mExportSelection, &numWaveTracks, &waveTracks);
Mixer *mixer = new Mixer(numWaveTracks, waveTracks,
tracks->GetTimeTrack(),
m_t0, m_t1,
info.channels, maxBlockLen, true,
mInRate, format);
while(!cancelling) {
sampleCount numSamples = mixer->Process(maxBlockLen);
if (numSamples == 0)
break;
samplePtr mixed = mixer->GetBuffer();
if (format == int16Sample)
sf_writef_short(sf, (short *)mixed, numSamples);
else
sf_writef_float(sf, (float *)mixed, numSamples);
if (!progress && wxGetElapsedTime(false) > 500) {
wxString message;
if (mExportSelection)
message =
wxString::
Format(_("Exporting the selected audio as a %s file"),
(const char *) formatStr);
else
message =
wxString::
Format(_("Exporting the entire project as a %s file"),
(const char *) formatStr);
progress =
new wxProgressDialog(_("Export"),
message,
1000,
parent,
wxPD_CAN_ABORT |
wxPD_REMAINING_TIME | wxPD_AUTO_HIDE);
}
if (progress) {
int progressvalue = int (1000 * ((mixer->MixGetCurrentTime()-m_t0) /
(m_t1-m_t0)));
cancelling = !progress->Update(progressvalue);
}
}
delete mixer;
delete[] waveTracks;
err = sf_close(sf);
if (err) {
char buffer[1000];
sf_error_str(sf, buffer, 1000);
wxMessageBox(wxString::Format
(_("Error (file may not have been written): %s"),
buffer));
}
#ifdef __WXMAC__
FSSpec spec;
wxMacFilename2FSSpec(fName, &spec);
FInfo finfo;
if (FSpGetFInfo(&spec, &finfo) == noErr) {
finfo.fdType = sf_header_mactype(mSoundFileFormat & SF_FORMAT_TYPEMASK);
finfo.fdCreator = AUDACITY_CREATOR;
FSpSetFInfo(&spec, &finfo);
}
#endif
if (progress)
delete progress;
return true;
}
bool ExportFFmpeg::EncodeAudioFrame(int16_t *pFrame, int frameSize)
{
AVPacket pkt;
int nBytesToWrite = 0;
uint8_t * pRawSamples = NULL;
int nAudioFrameSizeOut = mEncAudioCodecCtx->frame_size * mEncAudioCodecCtx->channels * sizeof(int16_t);
if (mEncAudioCodecCtx->frame_size == 1) nAudioFrameSizeOut = mEncAudioEncodedBufSiz;
int ret;
nBytesToWrite = frameSize;
pRawSamples = (uint8_t*)pFrame;
#if FFMPEG_STABLE
FFmpegLibsInst->av_fifo_realloc(&mEncAudioFifo, FFmpegLibsInst->av_fifo_size(&mEncAudioFifo) + frameSize);
#else
FFmpegLibsInst->av_fifo_realloc2(mEncAudioFifo, FFmpegLibsInst->av_fifo_size(mEncAudioFifo) + frameSize);
#endif
// Put the raw audio samples into the FIFO.
#if FFMPEG_STABLE
ret = FFmpegLibsInst->av_fifo_generic_write(&mEncAudioFifo, pRawSamples, nBytesToWrite,NULL);
#else
ret = FFmpegLibsInst->av_fifo_generic_write(mEncAudioFifo, pRawSamples, nBytesToWrite,NULL);
#endif
wxASSERT(ret == nBytesToWrite);
// Read raw audio samples out of the FIFO in nAudioFrameSizeOut byte-sized groups to encode.
#if FFMPEG_STABLE
while ((ret = FFmpegLibsInst->av_fifo_size(&mEncAudioFifo)) >= nAudioFrameSizeOut)
{
ret = FFmpegLibsInst->av_fifo_read(&mEncAudioFifo, mEncAudioFifoOutBuf, nAudioFrameSizeOut);
#else
while ((ret = FFmpegLibsInst->av_fifo_size(mEncAudioFifo)) >= nAudioFrameSizeOut)
{
ret = FFmpegLibsInst->av_fifo_generic_read(mEncAudioFifo, mEncAudioFifoOutBuf, nAudioFrameSizeOut, NULL);
#endif
FFmpegLibsInst->av_init_packet(&pkt);
pkt.size = FFmpegLibsInst->avcodec_encode_audio(mEncAudioCodecCtx,
mEncAudioEncodedBuf, mEncAudioEncodedBufSiz, // out
(int16_t*)mEncAudioFifoOutBuf); // in
if (mEncAudioCodecCtx->frame_size == 1) { wxASSERT(pkt.size == mEncAudioEncodedBufSiz); }
if (pkt.size < 0)
{
wxLogMessage(wxT("FFmpeg : ERROR - Can't encode audio frame."));
return false;
}
// Rescale from the codec time_base to the AVStream time_base.
if (mEncAudioCodecCtx->coded_frame && mEncAudioCodecCtx->coded_frame->pts != int64_t(AV_NOPTS_VALUE))
pkt.pts = FFmpegLibsInst->av_rescale_q(mEncAudioCodecCtx->coded_frame->pts, mEncAudioCodecCtx->time_base, mEncAudioStream->time_base);
//wxLogMessage(wxT("FFmpeg : (%d) Writing audio frame with PTS: %lld."), mEncAudioCodecCtx->frame_number, pkt.pts);
pkt.stream_index = mEncAudioStream->index;
pkt.data = mEncAudioEncodedBuf;
pkt.flags |= PKT_FLAG_KEY;
// Write the encoded audio frame to the output file.
if ((ret = FFmpegLibsInst->av_interleaved_write_frame(mEncFormatCtx, &pkt)) != 0)
{
wxLogMessage(wxT("FFmpeg : ERROR - Failed to write audio frame to file."));
return false;
}
}
return true;
}
int ExportFFmpeg::Export(AudacityProject *project,
int channels, wxString fName,
bool selectionOnly, double t0, double t1, MixerSpec *mixerSpec, Tags *metadata, int subformat)
{
if (!CheckFFmpegPresence())
return false;
mChannels = channels;
// subformat index may not correspond directly to fmts[] index, convert it
mSubFormat = AdjustFormatIndex(subformat);
if (channels > ExportFFmpegOptions::fmts[mSubFormat].maxchannels)
{
wxLogMessage(wxT("Attempted to export %d channels, but max. channels = %d"),channels,ExportFFmpegOptions::fmts[mSubFormat].maxchannels);
wxMessageBox(wxString::Format(_("Attempted to export %d channels, but max. channels for selected output format is %d"),channels,ExportFFmpegOptions::fmts[mSubFormat].maxchannels),_("Error"));
return false;
}
mName = fName;
TrackList *tracks = project->GetTracks();
bool ret = true;
if (mSubFormat >= FMT_LAST) return false;
wxString shortname(ExportFFmpegOptions::fmts[mSubFormat].shortname);
if (mSubFormat == FMT_OTHER)
shortname = gPrefs->Read(wxT("/FileFormats/FFmpegFormat"),wxT("matroska"));
ret = Init(shortname.mb_str(),project, metadata);
if (!ret) return false;
int pcmBufferSize = 1024;
int numWaveTracks;
WaveTrack **waveTracks;
tracks->GetWaveTracks(selectionOnly, &numWaveTracks, &waveTracks);
Mixer *mixer = new Mixer(numWaveTracks, waveTracks,
tracks->GetTimeTrack(),
t0, t1,
channels, pcmBufferSize, true,
mSampleRate, int16Sample, true, mixerSpec);
delete [] waveTracks;
ProgressDialog *progress = new ProgressDialog(wxFileName(fName).GetName(),
selectionOnly ?
wxString::Format(_("Exporting selected audio as %s"), ExportFFmpegOptions::fmts[mSubFormat].description) :
wxString::Format(_("Exporting entire file as %s"), ExportFFmpegOptions::fmts[mSubFormat].description));
int updateResult = eProgressSuccess;
while(updateResult == eProgressSuccess) {
sampleCount pcmNumSamples = mixer->Process(pcmBufferSize);
if (pcmNumSamples == 0)
break;
short *pcmBuffer = (short *)mixer->GetBuffer();
EncodeAudioFrame(pcmBuffer,(pcmNumSamples)*sizeof(int16_t)*mChannels);
updateResult = progress->Update(mixer->MixGetCurrentTime()-t0, t1-t0);
}
delete progress;
delete mixer;
Finalize();
return updateResult;
}
void AddStringTagUTF8(char field[], int size, wxString value)
{
memset(field,0,size);
memcpy(field,value.ToUTF8(),(int)strlen(value.ToUTF8()) > size -1 ? size -1 : strlen(value.ToUTF8()));
}
int ExportFFmpeg::Export(AudacityProject *project,
int channels, const wxString &fName,
bool selectionOnly, double t0, double t1, MixerSpec *mixerSpec, const Tags *metadata, int subformat)
{
if (!CheckFFmpegPresence())
return false;
mChannels = channels;
// subformat index may not correspond directly to fmts[] index, convert it
mSubFormat = AdjustFormatIndex(subformat);
if (channels > ExportFFmpegOptions::fmts[mSubFormat].maxchannels)
{
wxMessageBox(
wxString::Format(
_("Attempted to export %d channels, but maximum number of channels for selected output format is %d"),
channels,
ExportFFmpegOptions::fmts[mSubFormat].maxchannels),
_("Error"));
return false;
}
mName = fName;
TrackList *tracks = project->GetTracks();
bool ret = true;
if (mSubFormat >= FMT_LAST) return false;
wxString shortname(ExportFFmpegOptions::fmts[mSubFormat].shortname);
if (mSubFormat == FMT_OTHER)
shortname = gPrefs->Read(wxT("/FileFormats/FFmpegFormat"),wxT("matroska"));
ret = Init(shortname.mb_str(),project, metadata, subformat);
if (!ret) return false;
int pcmBufferSize = 1024;
int numWaveTracks;
WaveTrack **waveTracks;
tracks->GetWaveTracks(selectionOnly, &numWaveTracks, &waveTracks);
Mixer *mixer = CreateMixer(numWaveTracks, waveTracks,
tracks->GetTimeTrack(),
t0, t1,
channels, pcmBufferSize, true,
mSampleRate, int16Sample, true, mixerSpec);
delete[] waveTracks;
int updateResult = eProgressSuccess;
{
ProgressDialog progress(wxFileName(fName).GetName(),
selectionOnly ?
wxString::Format(_("Exporting selected audio as %s"), ExportFFmpegOptions::fmts[mSubFormat].description) :
wxString::Format(_("Exporting entire file as %s"), ExportFFmpegOptions::fmts[mSubFormat].description));
while (updateResult == eProgressSuccess) {
sampleCount pcmNumSamples = mixer->Process(pcmBufferSize);
if (pcmNumSamples == 0)
break;
short *pcmBuffer = (short *)mixer->GetBuffer();
EncodeAudioFrame(pcmBuffer, (pcmNumSamples)*sizeof(int16_t)*mChannels);
updateResult = progress.Update(mixer->MixGetCurrentTime() - t0, t1 - t0);
}
}
delete mixer;
Finalize();
return updateResult;
}
//TODO-MB: wouldn't it make more sense to delete the time track after 'mix and render'?
bool MixAndRender(TrackList *tracks, TrackFactory *trackFactory,
double rate, sampleFormat format,
double startTime, double endTime,
WaveTrack **newLeft, WaveTrack **newRight)
{
// This function was formerly known as "Quick Mix".
WaveTrack **waveArray;
Track *t;
int numWaves = 0; /* number of wave tracks in the selection */
int numMono = 0; /* number of mono, centre-panned wave tracks in selection*/
bool mono = false; /* flag if output can be mono without loosing anything*/
bool oneinput = false; /* flag set to true if there is only one input track
(mono or stereo) */
int w;
TrackListIterator iter(tracks);
SelectedTrackListOfKindIterator usefulIter(Track::Wave, tracks);
// this only iterates tracks which are relevant to this function, i.e.
// selected WaveTracks. The tracklist is (confusingly) the list of all
// tracks in the project
t = iter.First();
while (t) {
if (t->GetSelected() && t->GetKind() == Track::Wave) {
numWaves++;
float pan = ((WaveTrack*)t)->GetPan();
if (t->GetChannel() == Track::MonoChannel && pan == 0)
numMono++;
}
t = iter.Next();
}
if (numMono == numWaves)
mono = true;
/* the next loop will do two things at once:
* 1. build an array of all the wave tracks were are trying to process
* 2. determine when the set of WaveTracks starts and ends, in case we
* need to work out for ourselves when to start and stop rendering.
*/
double mixStartTime = 0.0; /* start time of first track to start */
bool gotstart = false; // flag indicates we have found a start time
double mixEndTime = 0.0; /* end time of last track to end */
double tstart, tend; // start and end times for one track.
waveArray = new WaveTrack *[numWaves];
w = 0;
t = iter.First();
while (t) {
if (t->GetSelected() && t->GetKind() == Track::Wave) {
waveArray[w++] = (WaveTrack *) t;
tstart = t->GetStartTime();
tend = t->GetEndTime();
if (tend > mixEndTime)
mixEndTime = tend;
// try and get the start time. If the track is empty we will get 0,
// which is ambiguous because it could just mean the track starts at
// the beginning of the project, as well as empty track. The give-away
// is that an empty track also ends at zero.
if (tstart != tend) {
// we don't get empty tracks here
if (!gotstart) {
// no previous start, use this one unconditionally
mixStartTime = tstart;
gotstart = true;
} else if (tstart < mixStartTime)
mixStartTime = tstart; // have a start, only make it smaller
} // end if start and end are different
} // end if track is a selected WaveTrack.
/** @TODO: could we not use a SelectedTrackListOfKindIterator here? */
t = iter.Next();
}
/* create the destination track (new track) */
if ((numWaves == 1) || ((numWaves == 2) && (usefulIter.First()->GetLink() != NULL)))
oneinput = true;
// only one input track (either 1 mono or one linked stereo pair)
WaveTrack *mixLeft = trackFactory->NewWaveTrack(format, rate);
if (oneinput)
mixLeft->SetName(usefulIter.First()->GetName()); /* set name of output track to be the same as the sole input track */
else
mixLeft->SetName(_("Mix"));
mixLeft->SetOffset(mixStartTime);
WaveTrack *mixRight = 0;
if (mono) {
mixLeft->SetChannel(Track::MonoChannel);
}
else {
mixRight = trackFactory->NewWaveTrack(format, rate);
if (oneinput) {
if (usefulIter.First()->GetLink() != NULL) // we have linked track
mixLeft->SetName(usefulIter.First()->GetLink()->GetName()); /* set name to match input track's right channel!*/
else
mixLeft->SetName(usefulIter.First()->GetName()); /* set name to that of sole input channel */
}
else
mixRight->SetName(_("Mix"));
mixLeft->SetChannel(Track::LeftChannel);
mixRight->SetChannel(Track::RightChannel);
mixRight->SetOffset(mixStartTime);
mixLeft->SetLinked(true);
}
int maxBlockLen = mixLeft->GetIdealBlockSize();
// If the caller didn't specify a time range, use the whole range in which
// any input track had clips in it.
if (startTime == endTime) {
startTime = mixStartTime;
endTime = mixEndTime;
}
Mixer *mixer = new Mixer(numWaves, waveArray,
Mixer::WarpOptions(tracks->GetTimeTrack()),
startTime, endTime, mono ? 1 : 2, maxBlockLen, false,
rate, format);
::wxSafeYield();
ProgressDialog *progress = new ProgressDialog(_("Mix and Render"),
_("Mixing and rendering tracks"));
int updateResult = eProgressSuccess;
while(updateResult == eProgressSuccess) {
sampleCount blockLen = mixer->Process(maxBlockLen);
if (blockLen == 0)
break;
if (mono) {
samplePtr buffer = mixer->GetBuffer();
mixLeft->Append(buffer, format, blockLen);
}
else {
samplePtr buffer;
buffer = mixer->GetBuffer(0);
mixLeft->Append(buffer, format, blockLen);
buffer = mixer->GetBuffer(1);
mixRight->Append(buffer, format, blockLen);
}
updateResult = progress->Update(mixer->MixGetCurrentTime() - startTime, endTime - startTime);
}
delete progress;
mixLeft->Flush();
if (!mono)
mixRight->Flush();
if (updateResult == eProgressCancelled || updateResult == eProgressFailed)
{
delete mixLeft;
if (!mono)
delete mixRight;
} else {
*newLeft = mixLeft;
if (!mono)
*newRight = mixRight;
#if 0
int elapsedMS = wxGetElapsedTime();
double elapsedTime = elapsedMS * 0.001;
double maxTracks = totalTime / (elapsedTime / numWaves);
// Note: these shouldn't be translated - they're for debugging
// and profiling only.
printf(" Tracks: %d\n", numWaves);
printf(" Mix length: %f sec\n", totalTime);
printf("Elapsed time: %f sec\n", elapsedTime);
printf("Max number of tracks to mix in real time: %f\n", maxTracks);
#endif
}
delete[] waveArray;
delete mixer;
return (updateResult == eProgressSuccess || updateResult == eProgressStopped);
}
bool ExportOGG(AudacityProject *project,
bool stereo, wxString fName,
bool selectionOnly, double t0, double t1)
{
double rate = project->GetRate();
wxWindow *parent = project;
TrackList *tracks = project->GetTracks();
double quality = (gPrefs->Read("/FileFormats/OggExportQuality", 50)/(float)100.0);
wxLogNull logNo; // temporarily disable wxWindows error messages
bool cancelling = false;
int eos = 0;
wxFFile outFile(fName, "wb");
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, stereo ? 2 : 1, int(rate + 0.5), quality);
vorbis_comment_init(&comment);
// If we wanted to add comments, we would do it here
// 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);
}
wxProgressDialog *progress = NULL;
wxYield();
wxStartTimer();
int numWaveTracks;
WaveTrack **waveTracks;
tracks->GetWaveTracks(selectionOnly, &numWaveTracks, &waveTracks);
Mixer *mixer = new Mixer(numWaveTracks, waveTracks,
tracks->GetTimeTrack(),
t0, t1,
stereo? 2: 1, SAMPLES_PER_RUN, false,
rate, floatSample);
while(!cancelling && !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 {
float *left = (float *)mixer->GetBuffer(0);
memcpy(vorbis_buffer[0], left, sizeof(float)*SAMPLES_PER_RUN);
if(stereo) {
float *right = (float *)mixer->GetBuffer(1);
memcpy(vorbis_buffer[1], right, 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;
}
}
}
if(progress) {
int progressvalue = int (1000 * ((mixer->MixGetCurrentTime()-t0) /
(t1-t0)));
cancelling = !progress->Update(progressvalue);
}
else if(wxGetElapsedTime(false) > 500) {
wxString message = selectionOnly ?
_("Exporting the selected audio as Ogg Vorbis") :
_("Exporting the entire project as Ogg Vorbis");
progress = new wxProgressDialog(
_("Export"),
message,
1000,
parent,
wxPD_CAN_ABORT | wxPD_REMAINING_TIME | wxPD_AUTO_HIDE);
}
}
delete mixer;
ogg_stream_clear(&stream);
vorbis_block_clear(&block);
vorbis_dsp_clear(&dsp);
vorbis_info_clear(&info);
outFile.Close();
if(progress)
delete progress;
return true;
}
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 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;
}