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-rw-r--r--muse/muse/node.cpp311
1 files changed, 248 insertions, 63 deletions
diff --git a/muse/muse/node.cpp b/muse/muse/node.cpp
index 4928420f..178c87cc 100644
--- a/muse/muse/node.cpp
+++ b/muse/muse/node.cpp
@@ -177,7 +177,7 @@ void AudioTrack::updateInternalSoloStates()
const RouteList* rl = inRoutes();
for(ciRoute ir = rl->begin(); ir != rl->end(); ++ir)
{
- if(ir->type == TRACK_ROUTE)
+ if(ir->type == Route::TRACK_ROUTE)
ir->track->updateInternalSoloStates();
}
}
@@ -186,7 +186,7 @@ void AudioTrack::updateInternalSoloStates()
const RouteList* rl = outRoutes();
for(ciRoute ir = rl->begin(); ir != rl->end(); ++ir)
{
- if(ir->type == TRACK_ROUTE)
+ if(ir->type == Route::TRACK_ROUTE)
ir->track->updateInternalSoloStates();
}
}
@@ -243,7 +243,7 @@ void AudioTrack::updateSoloStates(bool noDec)
const RouteList* rl = inRoutes();
for(ciRoute ir = rl->begin(); ir != rl->end(); ++ir)
{
- if(ir->type == TRACK_ROUTE)
+ if(ir->type == Route::TRACK_ROUTE)
ir->track->updateInternalSoloStates();
}
}
@@ -252,7 +252,7 @@ void AudioTrack::updateSoloStates(bool noDec)
const RouteList* rl = outRoutes();
for(ciRoute ir = rl->begin(); ir != rl->end(); ++ir)
{
- if(ir->type == TRACK_ROUTE)
+ if(ir->type == Route::TRACK_ROUTE)
ir->track->updateInternalSoloStates();
}
}
@@ -280,7 +280,8 @@ void Track::setOff(bool val)
// copyData
//---------------------------------------------------------
-void AudioTrack::copyData(unsigned pos, int dstChannels, unsigned nframes, float** dstBuffer)
+//void AudioTrack::copyData(unsigned pos, int dstChannels, unsigned nframes, float** dstBuffer)
+void AudioTrack::copyData(unsigned pos, int dstChannels, int srcStartChan, int srcChannels, unsigned nframes, float** dstBuffer)
{
//Changed by T356. 12/12/09.
// Overhaul and streamline to eliminate multiple processing during one process loop.
@@ -288,9 +289,14 @@ void AudioTrack::copyData(unsigned pos, int dstChannels, unsigned nframes, float
// Make better use of AudioTrack::outBuffers as a post-effect pre-volume cache system for multiple calls here during processing.
// Previously only WaveTrack used them. (Changed WaveTrack as well).
- int srcChannels = channels();
+ if(srcStartChan == -1)
+ srcStartChan = 0;
+
+ int srcChans = (srcChannels == -1) ? channels() : srcChannels;
+ int srcTotalOutChans = totalOutChannels();
+ if(channels() == 1)
+ srcTotalOutChans = 1;
- //Added by Tim. p3.3.16
#ifdef NODE_DEBUG
printf("MusE: AudioTrack::copyData name:%s processed:%d\n", name().latin1(), processed());
#endif
@@ -304,7 +310,11 @@ void AudioTrack::copyData(unsigned pos, int dstChannels, unsigned nframes, float
int i;
- float* buffer[srcChannels];
+ // p3.3.38
+ //float* buffer[srcChannels];
+ float* buffer[srcTotalOutChans];
+
+
//float data[nframes * srcChannels];
//for(i = 0; i < srcChannels; ++i)
// buffer[i] = data + i * nframes;
@@ -315,7 +325,7 @@ void AudioTrack::copyData(unsigned pos, int dstChannels, unsigned nframes, float
double _pan = pan();
vol[0] = _volume * (1.0 - _pan);
vol[1] = _volume * (1.0 + _pan);
- float meter[srcChannels];
+ float meter[srcChans];
// Have we been here already during this process cycle?
if(processed())
@@ -330,7 +340,10 @@ void AudioTrack::copyData(unsigned pos, int dstChannels, unsigned nframes, float
if(_haveData)
{
// Point the input buffers at our local cached 'pre-volume' buffers. They need processing, so continue on after.
- for(i = 0; i < srcChannels; ++i)
+ //for(i = 0; i < srcChannels; ++i)
+ // buffer[i] = outBuffers[i];
+ // p3.3.38
+ for(i = 0; i < srcTotalOutChans; ++i)
buffer[i] = outBuffers[i];
}
else
@@ -354,8 +367,12 @@ void AudioTrack::copyData(unsigned pos, int dstChannels, unsigned nframes, float
// First time here during this process cycle.
// Point the input buffers at a temporary stack buffer.
- float data[nframes * srcChannels];
- for(i = 0; i < srcChannels; ++i)
+ //float data[nframes * srcChannels];
+ //for(i = 0; i < srcChannels; ++i)
+ // buffer[i] = data + i * nframes;
+ // p3.3.38
+ float data[nframes * srcTotalOutChans];
+ for(i = 0; i < srcTotalOutChans; ++i)
buffer[i] = data + i * nframes;
// getData can use the supplied buffers, or change buffer to point to its own local buffers or Jack buffers etc.
@@ -363,9 +380,10 @@ void AudioTrack::copyData(unsigned pos, int dstChannels, unsigned nframes, float
// p3.3.29 1/27/10 Don't do any processing at all if off. Whereas, mute needs to be ready for action at all times,
// so still call getData before it. Off is NOT meant to be toggled rapidly, but mute is !
//if(!getData(pos, srcChannels, nframes, buffer) || off() || (isMute() && !_prefader))
- if(off() || !getData(pos, srcChannels, nframes, buffer) || (isMute() && !_prefader))
+ //if(off() || !getData(pos, srcChannels, nframes, buffer) || (isMute() && !_prefader))
+ // p3.3.38
+ if(off() || !getData(pos, srcTotalOutChans, nframes, buffer) || (isMute() && !_prefader))
{
- //Added by Tim. p3.3.16
#ifdef NODE_DEBUG
printf("MusE: AudioTrack::copyData name:%s dstChannels:%d zeroing buffers\n", name().latin1(), dstChannels);
#endif
@@ -383,7 +401,7 @@ void AudioTrack::copyData(unsigned pos, int dstChannels, unsigned nframes, float
memset(dstBuffer[i], 0, sizeof(float) * nframes);
}
- for(i = 0; i < srcChannels; ++i)
+ for(i = 0; i < srcChans; ++i)
{
//_meter[i] = 0;
_meter[i] = 0.0;
@@ -411,7 +429,7 @@ void AudioTrack::copyData(unsigned pos, int dstChannels, unsigned nframes, float
// apply plugin chain
//---------------------------------------------------
- _efxPipe->apply(srcChannels, nframes, buffer);
+ _efxPipe->apply(srcChans, nframes, buffer);
//---------------------------------------------------
// aux sends
@@ -429,9 +447,9 @@ void AudioTrack::copyData(unsigned pos, int dstChannels, unsigned nframes, float
AudioAux* a = (AudioAux*)((*al)[k]);
float** dst = a->sendBuffer();
int auxChannels = a->channels();
- if((srcChannels ==1 && auxChannels==1) || srcChannels == 2)
+ if((srcChans ==1 && auxChannels==1) || srcChans == 2)
{
- for(int ch = 0; ch < srcChannels; ++ch)
+ for(int ch = 0; ch < srcChans; ++ch)
{
float* db = dst[ch % a->channels()]; // no matter whether there's one or two dst buffers
float* sb = buffer[ch];
@@ -439,7 +457,7 @@ void AudioTrack::copyData(unsigned pos, int dstChannels, unsigned nframes, float
*db++ += (*sb++ * m * vol[ch]); // add to mix
}
}
- else if(srcChannels==1 && auxChannels==2) // copy mono to both channels
+ else if(srcChans==1 && auxChannels==2) // copy mono to both channels
{
for(int ch = 0; ch < auxChannels; ++ch)
{
@@ -458,7 +476,7 @@ void AudioTrack::copyData(unsigned pos, int dstChannels, unsigned nframes, float
if(_prefader)
{
- for(i = 0; i < srcChannels; ++i)
+ for(i = 0; i < srcChans; ++i)
{
float* p = buffer[i];
meter[i] = 0.0;
@@ -514,7 +532,10 @@ void AudioTrack::copyData(unsigned pos, int dstChannels, unsigned nframes, float
// 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 < srcChannels; ++i)
+ //for(i = 0; i < srcChannels; ++i)
+ // AL::dsp->cpy(outBuffers[i], buffer[i], nframes);
+ // p3.3.38
+ for(i = 0; i < srcTotalOutChans; ++i)
AL::dsp->cpy(outBuffers[i], buffer[i], nframes);
}
@@ -522,18 +543,43 @@ void AudioTrack::copyData(unsigned pos, int dstChannels, unsigned nframes, float
_haveData = true;
}
+ // Sanity check. Is source starting channel out of range? Just zero and return.
+ if(srcStartChan >= srcTotalOutChans)
+ {
+ unsigned int q;
+ for(i = 0; i < dstChannels; ++i)
+ {
+ if(config.useDenormalBias)
+ {
+ for(q = 0; q < nframes; q++)
+ dstBuffer[i][q] = denormalBias;
+ }
+ else
+ memset(dstBuffer[i], 0, sizeof(float) * nframes);
+ }
+ _processed = true;
+ return;
+ }
+ // Force a source range to fit actual available total out channels.
+ if((srcStartChan + srcChans) > srcTotalOutChans)
+ srcChans = srcTotalOutChans - srcStartChan;
+
//---------------------------------------------------
// apply volume
// postfader metering
//---------------------------------------------------
- if(srcChannels == dstChannels)
+
+ if(srcChans == dstChannels)
{
if(_prefader)
{
for(int c = 0; c < dstChannels; ++c)
{
- float* sp = buffer[c];
+ // p3.3.38
+ //float* sp = buffer[c];
+ float* sp = buffer[c + srcStartChan];
+
float* dp = dstBuffer[c];
for(unsigned k = 0; k < nframes; ++k)
*dp++ = (*sp++ * vol[c]);
@@ -544,7 +590,11 @@ void AudioTrack::copyData(unsigned pos, int dstChannels, unsigned nframes, float
for(int c = 0; c < dstChannels; ++c)
{
meter[c] = 0.0;
- float* sp = buffer[c];
+
+ // p3.3.38
+ //float* sp = buffer[c];
+ float* sp = buffer[c + srcStartChan];
+
float* dp = dstBuffer[c];
//printf("2 dstBuffer[c]=%d\n",long(dstBuffer[c]));
for(unsigned k = 0; k < nframes; ++k)
@@ -562,9 +612,12 @@ void AudioTrack::copyData(unsigned pos, int dstChannels, unsigned nframes, float
}
}
}
- else if(srcChannels == 1 && dstChannels == 2)
+ else if(srcChans == 1 && dstChannels == 2)
{
- float* sp = buffer[0];
+ // p3.3.38
+ //float* sp = buffer[0];
+ float* sp = buffer[srcStartChan];
+
if(_prefader)
{
for(int c = 0; c < dstChannels; ++c)
@@ -592,10 +645,14 @@ void AudioTrack::copyData(unsigned pos, int dstChannels, unsigned nframes, float
_peak[0] = _meter[0];
}
}
- else if(srcChannels == 2 && dstChannels == 1)
+ else if(srcChans == 2 && dstChannels == 1)
{
- float* sp1 = buffer[0];
- float* sp2 = buffer[1];
+ // p3.3.38
+ //float* sp1 = buffer[0];
+ //float* sp2 = buffer[1];
+ float* sp1 = buffer[srcStartChan];
+ float* sp2 = buffer[srcStartChan + 1];
+
if(_prefader)
{
float* dp = dstBuffer[0];
@@ -637,7 +694,8 @@ void AudioTrack::copyData(unsigned pos, int dstChannels, unsigned nframes, float
// addData
//---------------------------------------------------------
-void AudioTrack::addData(unsigned pos, int dstChannels, unsigned nframes, float** dstBuffer)
+//void AudioTrack::addData(unsigned pos, int dstChannels, unsigned nframes, float** dstBuffer)
+void AudioTrack::addData(unsigned pos, int dstChannels, int srcStartChan, int srcChannels, unsigned nframes, float** dstBuffer)
{
//Changed by T356. 12/12/09.
// Overhaul and streamline to eliminate multiple processing during one process loop.
@@ -656,7 +714,13 @@ void AudioTrack::addData(unsigned pos, int dstChannels, unsigned nframes, float*
return;
}
- int srcChannels = channels();
+ if(srcStartChan == -1)
+ srcStartChan = 0;
+
+ int srcChans = (srcChannels == -1) ? channels() : srcChannels;
+ int srcTotalOutChans = totalOutChannels();
+ if(channels() == 1)
+ srcTotalOutChans = 1;
// Special consideration for metronome: It is not part of the track list,
// and it has no in or out routes, yet multiple output tracks may call addData on it !
@@ -666,7 +730,10 @@ void AudioTrack::addData(unsigned pos, int dstChannels, unsigned nframes, float*
int i;
- float* buffer[srcChannels];
+ // p3.3.38
+ //float* buffer[srcChannels];
+ float* buffer[srcTotalOutChans];
+
//float data[nframes * srcChannels];
//for (i = 0; i < srcChannels; ++i)
// buffer[i] = data + i * nframes;
@@ -677,7 +744,7 @@ void AudioTrack::addData(unsigned pos, int dstChannels, unsigned nframes, float*
double _pan = pan();
vol[0] = _volume * (1.0 - _pan);
vol[1] = _volume * (1.0 + _pan);
- float meter[srcChannels];
+ float meter[srcChans];
// Have we been here already during this process cycle?
if(processed())
@@ -692,7 +759,10 @@ void AudioTrack::addData(unsigned pos, int dstChannels, unsigned nframes, float*
if(_haveData)
{
// Point the input buffers at our local cached 'pre-volume' buffers. They need processing, so continue on after.
- for(i = 0; i < srcChannels; ++i)
+ //for(i = 0; i < srcChannels; ++i)
+ // buffer[i] = outBuffers[i];
+ // p3.3.38
+ for(i = 0; i < srcTotalOutChans; ++i)
buffer[i] = outBuffers[i];
}
else
@@ -704,16 +774,23 @@ void AudioTrack::addData(unsigned pos, int dstChannels, unsigned nframes, float*
// First time here during this process cycle.
// Point the input buffers at a temporary stack buffer.
- float data[nframes * srcChannels];
- for(i = 0; i < srcChannels; ++i)
- buffer[i] = data + i * nframes;
+ //float data[nframes * srcChannels];
+ //for(i = 0; i < srcChannels; ++i)
+ // buffer[i] = data + i * nframes;
+ // p3.3.38
+ float data[nframes * srcTotalOutChans];
+ for(i = 0; i < srcTotalOutChans; ++i)
+ buffer[i] = data + i * nframes;
+
// getData can use the supplied buffers, or change buffer to point to its own local buffers or Jack buffers etc.
// For ex. if this is an audio input, Jack will set the pointers for us.
- if(!getData(pos, srcChannels, nframes, buffer))
+ //if(!getData(pos, srcChannels, nframes, buffer))
+ // p3.3.38
+ if(!getData(pos, srcTotalOutChans, nframes, buffer))
{
// No data was available. Nothing to add, but zero our local buffers and the meters.
- for(i = 0; i < srcChannels; ++i)
+ for(i = 0; i < srcChans; ++i)
{
// If we're using local buffers, we must zero them so that the next thing requiring them
// during this process cycle will see zeros.
@@ -743,7 +820,7 @@ void AudioTrack::addData(unsigned pos, int dstChannels, unsigned nframes, float*
// apply plugin chain
//---------------------------------------------------
- _efxPipe->apply(srcChannels, nframes, buffer);
+ _efxPipe->apply(srcChans, nframes, buffer);
//---------------------------------------------------
// aux sends
@@ -761,9 +838,9 @@ void AudioTrack::addData(unsigned pos, int dstChannels, unsigned nframes, float*
AudioAux* a = (AudioAux*)((*al)[k]);
float** dst = a->sendBuffer();
int auxChannels = a->channels();
- if((srcChannels ==1 && auxChannels==1) || srcChannels==2)
+ if((srcChans ==1 && auxChannels==1) || srcChans==2)
{
- for(int ch = 0; ch < srcChannels; ++ch)
+ for(int ch = 0; ch < srcChans; ++ch)
{
float* db = dst[ch % a->channels()];
float* sb = buffer[ch];
@@ -771,7 +848,7 @@ void AudioTrack::addData(unsigned pos, int dstChannels, unsigned nframes, float*
*db++ += (*sb++ * m * vol[ch]); // add to mix
}
}
- else if(srcChannels == 1 && auxChannels == 2)
+ else if(srcChans == 1 && auxChannels == 2)
{
for(int ch = 0; ch < auxChannels; ++ch)
{
@@ -790,7 +867,7 @@ void AudioTrack::addData(unsigned pos, int dstChannels, unsigned nframes, float*
if(_prefader)
{
- for(i = 0; i < srcChannels; ++i)
+ for(i = 0; i < srcChans; ++i)
{
float* p = buffer[i];
meter[i] = 0.0;
@@ -835,7 +912,10 @@ void AudioTrack::addData(unsigned pos, int dstChannels, unsigned nframes, float*
// 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 < srcChannels; ++i)
+ //for(i = 0; i < srcChannels; ++i)
+ // AL::dsp->cpy(outBuffers[i], buffer[i], nframes);
+ // p3.3.38
+ for(i = 0; i < srcTotalOutChans; ++i)
AL::dsp->cpy(outBuffers[i], buffer[i], nframes);
}
@@ -843,18 +923,42 @@ void AudioTrack::addData(unsigned pos, int dstChannels, unsigned nframes, float*
_haveData = true;
}
+ // Sanity check. Is source starting channel out of range? Just zero and return.
+ if(srcStartChan >= srcTotalOutChans)
+ {
+ unsigned int q;
+ for(i = 0; i < dstChannels; ++i)
+ {
+ if(config.useDenormalBias)
+ {
+ for(q = 0; q < nframes; q++)
+ dstBuffer[i][q] = denormalBias;
+ }
+ else
+ memset(dstBuffer[i], 0, sizeof(float) * nframes);
+ }
+ _processed = true;
+ return;
+ }
+ // Force a source range to fit actual available total out channels.
+ if((srcStartChan + srcChans) > srcTotalOutChans)
+ srcChans = srcTotalOutChans - srcStartChan;
+
//---------------------------------------------------
// apply volume
// postfader metering
//---------------------------------------------------
- if(srcChannels == dstChannels)
+ if(srcChans == dstChannels)
{
if(_prefader)
{
for(int c = 0; c < dstChannels; ++c)
{
- float* sp = buffer[c];
+ // p3.3.38
+ //float* sp = buffer[c];
+ float* sp = buffer[c + srcStartChan];
+
float* dp = dstBuffer[c];
for(unsigned k = 0; k < nframes; ++k)
*dp++ += (*sp++ * vol[c]);
@@ -865,7 +969,10 @@ void AudioTrack::addData(unsigned pos, int dstChannels, unsigned nframes, float*
for(int c = 0; c < dstChannels; ++c)
{
meter[c] = 0.0;
- float* sp = buffer[c];
+ // p3.3.38
+ //float* sp = buffer[c];
+ float* sp = buffer[c + srcStartChan];
+
float* dp = dstBuffer[c];
for(unsigned k = 0; k < nframes; ++k)
{
@@ -882,21 +989,24 @@ void AudioTrack::addData(unsigned pos, int dstChannels, unsigned nframes, float*
}
}
}
- else if(srcChannels == 1 && dstChannels == 2)
+ else if(srcChans == 1 && dstChannels == 2)
{
+ // p3.3.38
+ float* sp = buffer[srcStartChan];
+
if(_prefader)
{
for(int c = 0; c < dstChannels; ++c)
{
float* dp = dstBuffer[c];
- float* sp = buffer[0];
+ //float* sp = buffer[0];
for(unsigned k = 0; k < nframes; ++k)
*dp++ += (*sp++ * vol[c]);
}
}
else
{
- float* sp = buffer[0];
+ //float* sp = buffer[0];
meter[0] = 0.0;
for(unsigned k = 0; k < nframes; ++k)
{
@@ -913,10 +1023,14 @@ void AudioTrack::addData(unsigned pos, int dstChannels, unsigned nframes, float*
_peak[0] = _meter[0];
}
}
- else if(srcChannels == 2 && dstChannels == 1)
+ else if(srcChans == 2 && dstChannels == 1)
{
- float* sp1 = buffer[0];
- float* sp2 = buffer[1];
+ // p3.3.38
+ //float* sp1 = buffer[0];
+ //float* sp2 = buffer[1];
+ float* sp1 = buffer[srcStartChan];
+ float* sp2 = buffer[srcStartChan + 1];
+
if(_prefader)
{
float* dp = dstBuffer[0];
@@ -1096,7 +1210,6 @@ bool AudioTrack::getData(unsigned pos, int channels, unsigned nframes, float** b
RouteList* rl = inRoutes();
- // Added by Tim. p3.3.16
#ifdef NODE_DEBUG
printf("AudioTrack::getData name:%s inRoutes:%d\n", name().latin1(), rl->size());
#endif
@@ -1105,20 +1218,37 @@ bool AudioTrack::getData(unsigned pos, int channels, unsigned nframes, float** b
if (ir == rl->end())
return false;
- // Added by Tim. p3.3.16
+ if(ir->track->isMidiTrack())
+ return false;
+
#ifdef NODE_DEBUG
printf(" calling copyData on %s...\n", ir->track->name().latin1());
#endif
- ir->track->copyData(pos, channels, nframes, buffer);
+ // p3.3.38
+ //((AudioTrack*)ir->track)->copyData(pos, channels, nframes, buffer);
+ ((AudioTrack*)ir->track)->copyData(pos, channels,
+ //(ir->track->type() == Track::AUDIO_SOFTSYNTH && ir->channel != -1) ? ir->channel : 0,
+ ir->channel,
+ ir->channels,
+ nframes, buffer);
+
++ir;
for (; ir != rl->end(); ++ir) {
- // Added by Tim. p3.3.16
#ifdef NODE_DEBUG
printf(" calling addData on %s...\n", ir->track->name().latin1());
#endif
- ir->track->addData(pos, channels, nframes, buffer);
+ if(ir->track->isMidiTrack())
+ continue;
+
+ // p3.3.38
+ //((AudioTrack*)ir->track)->addData(pos, channels, nframes, buffer);
+ ((AudioTrack*)ir->track)->addData(pos, channels,
+ //(ir->track->type() == Track::AUDIO_SOFTSYNTH && ir->channel != -1) ? ir->channel : 0,
+ ir->channel,
+ ir->channels,
+ nframes, buffer);
}
return true;
}
@@ -1368,7 +1498,7 @@ void AudioOutput::processInit(unsigned nframes)
//---------------------------------------------------------
void AudioOutput::process(unsigned pos, unsigned offset, unsigned n)
- {
+{
//Added by Tim. p3.3.16
#ifdef NODE_DEBUG
printf("MusE: AudioOutput::process name:%s processed:%d\n", name().latin1(), processed());
@@ -1377,8 +1507,11 @@ void AudioOutput::process(unsigned pos, unsigned offset, unsigned n)
for (int i = 0; i < _channels; ++i) {
buffer1[i] = buffer[i] + offset;
}
- copyData(pos, _channels, n, buffer1);
- }
+
+ // p3.3.38
+ //copyData(pos, _channels, n, buffer1);
+ copyData(pos, _channels, -1, -1, n, buffer1);
+}
//---------------------------------------------------------
// silence
@@ -1427,7 +1560,9 @@ void AudioOutput::processWrite()
printf("MusE: AudioOutput::processWrite Calling metronome->addData frame:%u channels:%d frames:%lu\n", audio->pos().frame(), _channels, _nframes);
#endif
- metronome->addData(audio->pos().frame(), _channels, _nframes, buffer);
+ // p3.3.38
+ //metronome->addData(audio->pos().frame(), _channels, _nframes, buffer);
+ metronome->addData(audio->pos().frame(), _channels, -1, -1, _nframes, buffer);
}
}
//---------------------------------------------------------
@@ -1606,3 +1741,53 @@ void AudioTrack::setChannels(int n)
_efxPipe->setChannels(n);
}
+//---------------------------------------------------------
+// setTotalOutChannels
+//---------------------------------------------------------
+
+void AudioTrack::setTotalOutChannels(int num)
+{
+ if(num == _totalOutChannels)
+ return;
+
+ 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;
+ for(int i = 0; i < chans; ++i)
+ {
+ if(outBuffers[i])
+ free(outBuffers[i]);
+ }
+ delete[] outBuffers;
+
+ _totalOutChannels = num;
+ chans = num;
+ // Number of allocated buffers is always MAX_CHANNELS or more, even if _totalOutChannels is less.
+ if(chans < MAX_CHANNELS)
+ chans = MAX_CHANNELS;
+
+ outBuffers = new float*[chans];
+ for (int i = 0; i < chans; ++i)
+ posix_memalign((void**)&outBuffers[i], 16, sizeof(float) * segmentSize);
+
+ chans = num;
+ // Limit the actual track (meters, copying etc, all 'normal' operation) to two-channel stereo.
+ if(chans > MAX_CHANNELS)
+ chans = MAX_CHANNELS;
+
+ setChannels(chans);
+}
+
+//---------------------------------------------------------
+// setTotalInChannels
+//---------------------------------------------------------
+
+void AudioTrack::setTotalInChannels(int num)
+{
+ if(num == _totalInChannels)
+ return;
+
+ _totalInChannels = num;
+}
+
generated by cgit v1.2.3 (git 2.39.1) at 2024-09-14 10:31:13 +0000