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-rw-r--r--muse2/muse/node.cpp159
1 files changed, 8 insertions, 151 deletions
diff --git a/muse2/muse/node.cpp b/muse2/muse/node.cpp
index 20440b45..be72367f 100644
--- a/muse2/muse/node.cpp
+++ b/muse2/muse/node.cpp
@@ -392,7 +392,6 @@ void AudioTrack::copyData(unsigned pos, int dstChannels, int srcStartChan, int s
// and it has no in or out routes, yet multiple output tracks may call addData on it !
// We can't tell how many output tracks call it, so we can only assume there might be more than one.
// Not strictly necessary here because only addData is ever called, but just to be consistent...
- //bool usedirectbuf = ((outRoutes()->size() <= 1) || (type() == AUDIO_OUTPUT)) && (this != metronome);
int i;
@@ -468,23 +467,8 @@ void AudioTrack::copyData(unsigned pos, int dstChannels, int srcStartChan, int s
_efxPipe->apply(0, nframes, 0); // Just process controls only, not audio (do not 'run').
for(i = 0; i < trackChans; ++i)
- {
_meter[i] = 0.0;
- /*if(!usedirectbuf)
- {
- if(MusEGlobal::config.useDenormalBias)
- {
- for(q = 0; q < nframes; ++q)
- outBuffers[i][q] = MusEGlobal::denormalBias;
- }
- else
- memset(outBuffers[i], 0, sizeof(float) * nframes);
- } */
- }
- //_haveData = false;
- //_processed = true;
- //_isProcessing = false; // Unblock.
return;
}
@@ -520,7 +504,6 @@ void AudioTrack::copyData(unsigned pos, int dstChannels, int srcStartChan, int s
// apply plugin chain
//---------------------------------------------------
- //fprintf(stderr, "AudioTrack::copyData %s efx apply srcChans:%d\n", name().toLatin1().constData(), srcChans);
_efxPipe->apply(trackChans, nframes, buffer);
//---------------------------------------------------
@@ -598,22 +581,6 @@ void AudioTrack::copyData(unsigned pos, int dstChannels, int srcStartChan, int s
memset(dstBuffer[i], 0, sizeof(float) * nframes);
}
- /*
- if(!usedirectbuf)
- {
- for(i = 0; i < srcChannels; ++i)
- {
- if(MusEGlobal::config.useDenormalBias)
- {
- for(q = 0; q < nframes; ++q)
- outBuffers[i][q] = MusEGlobal::denormalBias;
- }
- else
- memset(outBuffers[i], 0, sizeof(float) * nframes);
- }
- } */
-
-
if(!_prefader)
for(i = 0; i < trackChans; ++i) // Must process ALL channels, even if unconnected. Only max 2 channels.
_meter[i] = 0.0;
@@ -622,11 +589,8 @@ void AudioTrack::copyData(unsigned pos, int dstChannels, int srcStartChan, int s
}
// If we're using local cached 'pre-volume' buffers, copy the input buffers (as they are right now: post-effect pre-volume) back to them.
- //if(!usedirectbuf)
- {
- for(i = 0; i < srcTotalOutChans; ++i)
- AL::dsp->cpy(outBuffers[i], buffer[i], nframes);
- }
+ for(i = 0; i < srcTotalOutChans; ++i)
+ AL::dsp->cpy(outBuffers[i], buffer[i], nframes);
// We have some data! Set to true.
_haveData = true;
@@ -688,7 +652,6 @@ void AudioTrack::copyData(unsigned pos, int dstChannels, int srcStartChan, int s
{
double v;
if(srcStartChan > 2 || _prefader) // Don't apply pan or volume to extra channels above 2. Or if prefader on.
- //v = _volume;
v = 1.0;
else
if(srcChans >= 2) // If 2 channels apply pan normally.
@@ -702,7 +665,6 @@ void AudioTrack::copyData(unsigned pos, int dstChannels, int srcStartChan, int s
float* sp = buffer[c + srcStartChan];
float* dp = dstBuffer[c];
for(unsigned k = 0; k < nframes; ++k)
- //*dp++ = (*sp++ * vol[c]);
*dp++ = (*sp++ * v);
}
}
@@ -712,7 +674,6 @@ void AudioTrack::copyData(unsigned pos, int dstChannels, int srcStartChan, int s
{
double v;
if(srcStartChan > 2 || _prefader) // Don't apply pan or volume to extra channels above 2. Or if prefader on.
- //v = _volume;
v = 1.0;
else
if(trackChans <= 1) // If track is mono apply pan.
@@ -723,21 +684,17 @@ void AudioTrack::copyData(unsigned pos, int dstChannels, int srcStartChan, int s
float* sp = buffer[srcStartChan];
float* dp = dstBuffer[c];
for(unsigned k = 0; k < nframes; ++k)
- //*dp++ = (*sp++ * vol[c]);
*dp++ = (*sp++ * v);
}
}
else if(srcChans == 2 && dstChannels == 1)
{
- //double v1 = (srcStartChan > 2 ? _volume : vol[srcStartChan]); //
- //double v2 = (srcStartChan > 2 ? _volume : vol[srcStartChan + 1]); //
double v1 = ((srcStartChan > 2 || _prefader) ? 1.0 : vol[srcStartChan]); // Don't apply pan or volume to extra channels above 2. Or if prefader on.
double v2 = ((srcStartChan > 2 || _prefader) ? 1.0 : vol[srcStartChan + 1]); //
float* dp = dstBuffer[0];
float* sp1 = buffer[srcStartChan];
float* sp2 = buffer[srcStartChan + 1];
for(unsigned k = 0; k < nframes; ++k)
- //*dp++ = (*sp1++ * vol[0] + *sp2++ * vol[1]);
*dp++ = (*sp1++ * v1 + *sp2++ * v2);
}
}
@@ -758,12 +715,6 @@ void AudioTrack::addData(unsigned pos, int dstChannels, int srcStartChan, int sr
printf("MusE: AudioTrack::addData name:%s processed:%d\n", name().toLatin1().constData(), processed());
#endif
- //if (off())
- //{
- // _processed = true;
- // return;
- //}
-
if(srcStartChan == -1)
srcStartChan = 0;
@@ -773,10 +724,9 @@ void AudioTrack::addData(unsigned pos, int dstChannels, int srcStartChan, int sr
if(channels() == 1)
srcTotalOutChans = 1;
- // Special consideration for metronome: It is not part of the track list,
+ // Special consideration for metronome: It is not part of the track list, DELETETHIS??
// and it has no in or out routes, yet multiple output tracks may call addData on it !
// We can't tell how many output tracks call it, so we can only assume there might be more than one.
- //bool usedirectbuf = ((outRoutes()->size() <= 1) || (type() == AUDIO_OUTPUT)) && (this != metronome);
int i;
@@ -789,7 +739,6 @@ void AudioTrack::addData(unsigned pos, int dstChannels, int srcStartChan, int sr
double _pan = pan();
vol[0] = _volume * (1.0 - _pan);
vol[1] = _volume * (1.0 + _pan);
- //float meter[srcChans];
float meter[trackChans];
// Have we been here already during this process cycle?
@@ -832,19 +781,7 @@ void AudioTrack::addData(unsigned pos, int dstChannels, int srcStartChan, int sr
_efxPipe->apply(0, nframes, 0); // Track is off. Just process controls only, not audio (do not 'run').
for(i = 0; i < trackChans; ++i)
- {
_meter[i] = 0.0;
- /*if(!usedirectbuf)
- {
- if(MusEGlobal::config.useDenormalBias)
- {
- for(q = 0; q < nframes; ++q)
- outBuffers[i][q] = MusEGlobal::denormalBias;
- }
- else
- memset(outBuffers[i], 0, sizeof(float) * nframes);
- } */
- }
return;
}
@@ -873,15 +810,9 @@ void AudioTrack::addData(unsigned pos, int dstChannels, int srcStartChan, int sr
//---------------------------------------------------
// apply plugin chain
//---------------------------------------------------
-
- //fprintf(stderr, "AudioTrack::addData %s efx apply srcChans:%d nframes:%ld %e %e %e %e\n",
- // name().toLatin1().constData(), srcChans, nframes, buffer[0][0], buffer[0][1], buffer[0][2], buffer[0][3]);
_efxPipe->apply(trackChans, nframes, buffer);
- //fprintf(stderr, "AudioTrack::addData after efx: %e %e %e %e\n",
- // buffer[0][0], buffer[0][1], buffer[0][2], buffer[0][3]);
-
//---------------------------------------------------
// aux sends
//---------------------------------------------------
@@ -945,23 +876,7 @@ void AudioTrack::addData(unsigned pos, int dstChannels, int srcStartChan, int sr
if(isMute())
{
- // If we're using local buffers, we must zero them.
- /* if(!usedirectbuf)
- {
- for(i = 0; i < srcChannels; ++i)
- {
- if(MusEGlobal::config.useDenormalBias)
- {
- for(unsigned int q = 0; q < nframes; ++q)
- outBuffers[i][q] = MusEGlobal::denormalBias;
- }
- else
- memset(outBuffers[i], 0, sizeof(float) * nframes);
- }
- } */
-
if(!_prefader)
- //for(i = 0; i < srcChans; ++i)
for(i = 0; i < trackChans; ++i)
_meter[i] = 0.0;
@@ -969,11 +884,8 @@ void AudioTrack::addData(unsigned pos, int dstChannels, int srcStartChan, int sr
}
// If we're using local cached 'pre-volume' buffers, copy the input buffers (as they are right now: post-effect pre-volume) back to them.
- //if(!usedirectbuf)
- {
- for(i = 0; i < srcTotalOutChans; ++i)
- AL::dsp->cpy(outBuffers[i], buffer[i], nframes);
- }
+ for(i = 0; i < srcTotalOutChans; ++i)
+ AL::dsp->cpy(outBuffers[i], buffer[i], nframes);
// We have some data! Set to true.
_haveData = true;
@@ -1035,7 +947,6 @@ void AudioTrack::addData(unsigned pos, int dstChannels, int srcStartChan, int sr
{
double v;
if(srcStartChan > 2 || _prefader) // Don't apply pan or volume to extra channels above 2. Or if prefader on.
- //v = _volume;
v = 1.0;
else
if(srcChans >= 2) // If 2 channels apply pan normally.
@@ -1049,7 +960,6 @@ void AudioTrack::addData(unsigned pos, int dstChannels, int srcStartChan, int sr
float* sp = buffer[c + srcStartChan];
float* dp = dstBuffer[c];
for(unsigned k = 0; k < nframes; ++k)
- //*dp++ += (*sp++ * vol[c]);
*dp++ += (*sp++ * v);
}
}
@@ -1059,7 +969,6 @@ void AudioTrack::addData(unsigned pos, int dstChannels, int srcStartChan, int sr
{
double v;
if(srcStartChan > 2 || _prefader) // Don't apply pan or volume to extra channels above 2. Or if prefader on.
- //v = _volume;
v = 1.0;
else
if(trackChans <= 1) // If track is mono apply pan.
@@ -1070,21 +979,17 @@ void AudioTrack::addData(unsigned pos, int dstChannels, int srcStartChan, int sr
float* sp = buffer[srcStartChan];
float* dp = dstBuffer[c];
for(unsigned k = 0; k < nframes; ++k)
- //*dp++ += (*sp++ * vol[c]);
*dp++ += (*sp++ * v);
}
}
else if(srcChans == 2 && dstChannels == 1)
{
- //double v1 = (srcStartChan > 2 ? _volume : vol[srcStartChan]); // Don't apply pan to extra channels above 2.
- //double v2 = (srcStartChan > 2 ? _volume : vol[srcStartChan + 1]); //
double v1 = ((srcStartChan > 2 || _prefader) ? 1.0 : vol[srcStartChan]); // Don't apply pan or volume to extra channels above 2. Or if prefader on.
double v2 = ((srcStartChan > 2 || _prefader) ? 1.0 : vol[srcStartChan + 1]); //
float* sp1 = buffer[srcStartChan];
float* sp2 = buffer[srcStartChan + 1];
float* dp = dstBuffer[0];
for(unsigned k = 0; k < nframes; ++k)
- //*dp++ += (*sp1++ * vol[0] + *sp2++ * vol[1]);
*dp++ += (*sp1++ * v1 + *sp2++ * v2);
}
}
@@ -1095,7 +1000,6 @@ void AudioTrack::addData(unsigned pos, int dstChannels, int srcStartChan, int sr
void AudioTrack::readVolume(Xml& xml)
{
- //int ch = 0;
for (;;) {
Xml::Token token = xml.parse();
switch (token) {
@@ -1122,6 +1026,7 @@ void AudioTrack::readVolume(Xml& xml)
}
}
+// DELETETHIS 56
// Removed by T356
// "recfile" tag not saved anymore
/*
@@ -1189,9 +1094,7 @@ void Track::setChannels(int n)
else
_channels = n;
for (int i = 0; i < _channels; ++i) {
- //_meter[i] = 0;
_meter[i] = 0.0;
- //_peak[i] = 0;
_peak[i] = 0.0;
}
}
@@ -1201,7 +1104,6 @@ void AudioInput::setChannels(int n)
{
if (n == _channels)
return;
-//was ist mit: void* jackPorts[MAX_CHANNELS];
AudioTrack::setChannels(n);
}
@@ -1250,12 +1152,10 @@ bool AudioTrack::getData(unsigned pos, int channels, unsigned nframes, float** b
#endif
((AudioTrack*)ir->track)->copyData(pos, channels,
- //(ir->track->type() == Track::AUDIO_SOFTSYNTH && ir->channel != -1) ? ir->channel : 0,
ir->channel,
ir->channels,
nframes, buffer);
- //fprintf(stderr, "AudioTrack::getData %s data: nframes:%ld %e %e %e %e\n", name().toLatin1().constData(), nframes, buffer[0][0], buffer[0][1], buffer[0][2], buffer[0][3]);
++ir;
for (; ir != rl->end(); ++ir) {
@@ -1286,7 +1186,6 @@ bool AudioInput::getData(unsigned, int channels, unsigned nframes, float** buffe
for (int ch = 0; ch < channels; ++ch)
{
void* jackPort = jackPorts[ch];
- //float* jackbuf = 0;
// Do not get buffers of unconnected client ports. Causes repeating leftover data, can be loud, or DC !
if (jackPort && MusEGlobal::audioDevice->connections(jackPort))
@@ -1302,15 +1201,12 @@ bool AudioInput::getData(unsigned, int channels, unsigned nframes, float** buffe
// their channel port buffers (if that's even possible) in order to determine if the buffer is shared,
// let's just copy always, for now shall we ?
float* jackbuf = MusEGlobal::audioDevice->getBuffer(jackPort, nframes);
- //memcpy(buffer[ch], jackbuf, nframes* sizeof(float));
AL::dsp->cpy(buffer[ch], jackbuf, nframes);
if (MusEGlobal::config.useDenormalBias)
{
for (unsigned int i=0; i < nframes; i++)
buffer[ch][i] += MusEGlobal::denormalBias;
- //fprintf(stderr, "AudioInput::getData %s Jack port %p efx apply channels:%d nframes:%ld %e %e %e %e\n",
- // name().toLatin1().constData(), jackPort, channels, nframes, buffer[0][0], buffer[0][1], buffer[0][2], buffer[0][3]);
}
}
else
@@ -1324,8 +1220,6 @@ bool AudioInput::getData(unsigned, int channels, unsigned nframes, float** buffe
{
memset(buffer[ch], 0, nframes * sizeof(float));
}
-
- // name().toLatin1().constData(), channels, nframes, buffer[0][0], buffer[0][1], buffer[0][2], buffer[0][3]);
}
}
return true;
@@ -1345,7 +1239,6 @@ void AudioInput::setName(const QString& s)
if (jackPorts[i])
MusEGlobal::audioDevice->setPortName(jackPorts[i], buffer);
else {
- //jackPorts[i] = MusEGlobal::audioDevice->registerInPort(buffer);
jackPorts[i] = MusEGlobal::audioDevice->registerInPort(buffer, false);
}
}
@@ -1359,7 +1252,6 @@ void AudioInput::setName(const QString& s)
void Track::resetMeter()
{
for (int i = 0; i < _channels; ++i)
- //_meter[i] = 0;
_meter[i] = 0.0;
}
@@ -1370,7 +1262,6 @@ void Track::resetMeter()
void Track::resetPeaks()
{
for (int i = 0; i < _channels; ++i)
- //_peak[i] = 0;
_peak[i] = 0.0;
_lastActivity = 0;
}
@@ -1454,8 +1345,6 @@ void AudioTrack::record()
unsigned pos = 0;
float* buffer[_channels];
- //printf("AudioTrack: record() fifo %p, count=%d\n", &fifo, fifo.getCount());
-
while(fifo.getCount()) {
if (fifo.get(_channels, MusEGlobal::segmentSize, buffer, &pos)) {
@@ -1509,8 +1398,6 @@ void AudioTrack::record()
if( (pos >= fr) && (!MusEGlobal::song->punchout() || (!MusEGlobal::song->loop() && pos < MusEGlobal::song->rPos().frame())) )
{
pos -= fr;
- //int position = _recFile->seek(0, SEEK_CUR);
- //printf("AudioTrack::record loopcnt:%d lframe:%d newpos:%d curpos:%d start:%d end:%d\n", MusEGlobal::audio->loopCount(), MusEGlobal::audio->loopFrame(), pos, position, MusEGlobal::audio->getStartRecordPos().frame(), MusEGlobal::audio->getEndRecordPos().frame());
_recFile->seek(pos, 0);
_recFile->write(_channels, buffer, MusEGlobal::segmentSize);
@@ -1600,8 +1487,6 @@ void AudioOutput::processWrite()
putFifo(_channels, _nframes, buffer);
}
}
- // Changed by Tim.
- //if (MusEGlobal::audioClickFlag && MusEGlobal::song->click()) {
if (sendMetronome() && MusEGlobal::audioClickFlag && MusEGlobal::song->click()) {
#ifdef METRONOME_DEBUG
@@ -1621,13 +1506,10 @@ void AudioOutput::setName(const QString& s)
for (int i = 0; i < channels(); ++i) {
char buffer[128];
snprintf(buffer, 128, "%s-%d", _name.toLatin1().constData(), i);
- if (jackPorts[i]) {
+ if (jackPorts[i])
MusEGlobal::audioDevice->setPortName(jackPorts[i], buffer);
- }
- else {
- //jackPorts[i] = MusEGlobal::audioDevice->registerOutPort(buffer);
+ else
jackPorts[i] = MusEGlobal::audioDevice->registerOutPort(buffer, false);
- }
}
}
@@ -1639,7 +1521,6 @@ void AudioOutput::setName(const QString& s)
Fifo::Fifo()
{
muse_atomic_init(&count);
- //nbuffer = FIFO_BUFFER;
nbuffer = MusEGlobal::fifoLength;
buffer = new FifoBuffer*[nbuffer];
for (int i = 0; i < nbuffer; ++i)
@@ -1652,10 +1533,7 @@ Fifo::~Fifo()
for (int i = 0; i < nbuffer; ++i)
{
if(buffer[i]->buffer)
- {
- //printf("Fifo::~Fifo freeing buffer\n");
free(buffer[i]->buffer);
- }
delete buffer[i];
}
@@ -1684,14 +1562,9 @@ bool Fifo::put(int segs, unsigned long samples, float** src, unsigned pos)
if (b->maxSize < n) {
if (b->buffer)
{
- // Changed by Tim. p3.3.15
- //delete[] b->buffer;
free(b->buffer);
- // p3.3.45
b->buffer = 0;
}
- // Changed by Tim. p3.3.15
- //b->buffer = new float[n];
posix_memalign((void**)&(b->buffer), 16, sizeof(float) * n);
if(!b->buffer)
{
@@ -1711,7 +1584,6 @@ bool Fifo::put(int segs, unsigned long samples, float** src, unsigned pos)
b->segs = segs;
b->pos = pos;
for (int i = 0; i < segs; ++i)
- //memcpy(b->buffer + i * samples, src[i], samples * sizeof(float));
AL::dsp->cpy(b->buffer + i * samples, src[i], samples);
add();
return false;
@@ -1779,15 +1651,10 @@ bool Fifo::getWriteBuffer(int segs, unsigned long samples, float** buf, unsigned
if (b->maxSize < n) {
if (b->buffer)
{
- // Changed by Tim. p3.3.15
- //delete[] b->buffer;
free(b->buffer);
- // p3.3.45
b->buffer = 0;
}
- // Changed by Tim. p3.3.15
- //b->buffer = new float[n];
posix_memalign((void**)&(b->buffer), 16, sizeof(float) * n);
if(!b->buffer)
{
@@ -1840,14 +1707,9 @@ void AudioTrack::setChannels(int n)
void AudioTrack::setTotalOutChannels(int num)
{
- //if(num == _totalOutChannels)
- // return;
- // p4.0.27 Fixes crash if file loaded with track channels less than synth channels.
int chans = _totalOutChannels;
if(num != chans)
{
-
- //int chans = _totalOutChannels;
// Number of allocated buffers is always MAX_CHANNELS or more, even if _totalOutChannels is less.
if(chans < MAX_CHANNELS)
chans = MAX_CHANNELS;
@@ -1870,11 +1732,6 @@ void AudioTrack::setTotalOutChannels(int num)
outBuffers = new float*[chans];
for (int i = 0; i < chans; ++i)
posix_memalign((void**)&outBuffers[i], 16, sizeof(float) * MusEGlobal::segmentSize);
-
- //chans = num;
- // Limit the actual track (meters, copying etc, all 'normal' operation) to two-channel stereo.
- //if(chans > MAX_CHANNELS)
- // chans = MAX_CHANNELS;
}
chans = num;
// Limit the actual track (meters, copying etc, all 'normal' operation) to two-channel stereo.
generated by cgit v1.2.3 (git 2.39.1) at 2024-07-03 02:51:16 +0000