summaryrefslogtreecommitdiff
path: root/muse2/muse/node.cpp
diff options
context:
space:
mode:
Diffstat (limited to 'muse2/muse/node.cpp')
-rw-r--r--muse2/muse/node.cpp93
1 files changed, 54 insertions, 39 deletions
diff --git a/muse2/muse/node.cpp b/muse2/muse/node.cpp
index da3874f0..8b9e44d8 100644
--- a/muse2/muse/node.cpp
+++ b/muse2/muse/node.cpp
@@ -4,6 +4,21 @@
// $Id: node.cpp,v 1.36.2.25 2009/12/20 05:00:35 terminator356 Exp $
//
// (C) Copyright 2000-2004 Werner Schweer (ws@seh.de)
+//
+// This program is free software; you can redistribute it and/or
+// modify it under the terms of the GNU General Public License
+// as published by the Free Software Foundation; version 2 of
+// the License, or (at your option) any later version.
+//
+// This program is distributed in the hope that it will be useful,
+// but WITHOUT ANY WARRANTY; without even the implied warranty of
+// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+// GNU General Public License for more details.
+//
+// You should have received a copy of the GNU General Public License
+// along with this program; if not, write to the Free Software
+// Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301, USA.
+//
//=========================================================
#include <cmath>
@@ -375,10 +390,10 @@ void AudioTrack::copyData(unsigned pos, int dstChannels, int srcStartChan, int s
// No data was available from a previous call during this process cycle. Zero the supplied buffers and just return.
for(i = 0; i < dstChannels; ++i)
{
- if(config.useDenormalBias)
+ if(MusEConfig::config.useDenormalBias)
{
for(unsigned int q = 0; q < nframes; ++q)
- dstBuffer[i][q] = denormalBias;
+ dstBuffer[i][q] = MusEGlobal::denormalBias;
}
else
memset(dstBuffer[i], 0, sizeof(float) * nframes);
@@ -416,10 +431,10 @@ void AudioTrack::copyData(unsigned pos, int dstChannels, int srcStartChan, int s
unsigned int q;
for(i = 0; i < dstChannels; ++i)
{
- if(config.useDenormalBias)
+ if(MusEConfig::config.useDenormalBias)
{
for(q = 0; q < nframes; ++q)
- dstBuffer[i][q] = denormalBias;
+ dstBuffer[i][q] = MusEGlobal::denormalBias;
}
else
memset(dstBuffer[i], 0, sizeof(float) * nframes);
@@ -433,10 +448,10 @@ void AudioTrack::copyData(unsigned pos, int dstChannels, int srcStartChan, int s
/*
if(!usedirectbuf)
{
- if(config.useDenormalBias)
+ if(MusEConfig::config.useDenormalBias)
{
for(q = 0; q < nframes; ++q)
- outBuffers[i][q] = denormalBias;
+ outBuffers[i][q] = MusEGlobal::denormalBias;
}
else
memset(outBuffers[i], 0, sizeof(float) * nframes);
@@ -525,10 +540,10 @@ void AudioTrack::copyData(unsigned pos, int dstChannels, int srcStartChan, int s
unsigned int q;
for(i = 0; i < dstChannels; ++i)
{
- if(config.useDenormalBias)
+ if(MusEConfig::config.useDenormalBias)
{
for(q = 0; q < nframes; q++)
- dstBuffer[i][q] = denormalBias;
+ dstBuffer[i][q] = MusEGlobal::denormalBias;
}
else
memset(dstBuffer[i], 0, sizeof(float) * nframes);
@@ -539,10 +554,10 @@ void AudioTrack::copyData(unsigned pos, int dstChannels, int srcStartChan, int s
{
for(i = 0; i < srcChannels; ++i)
{
- if(config.useDenormalBias)
+ if(MusEConfig::config.useDenormalBias)
{
for(q = 0; q < nframes; ++q)
- outBuffers[i][q] = denormalBias;
+ outBuffers[i][q] = MusEGlobal::denormalBias;
}
else
memset(outBuffers[i], 0, sizeof(float) * nframes);
@@ -575,10 +590,10 @@ void AudioTrack::copyData(unsigned pos, int dstChannels, int srcStartChan, int s
unsigned int q;
for(i = 0; i < dstChannels; ++i)
{
- if(config.useDenormalBias)
+ if(MusEConfig::config.useDenormalBias)
{
for(q = 0; q < nframes; q++)
- dstBuffer[i][q] = denormalBias;
+ dstBuffer[i][q] = MusEGlobal::denormalBias;
}
else
memset(dstBuffer[i], 0, sizeof(float) * nframes);
@@ -823,10 +838,10 @@ void AudioTrack::addData(unsigned pos, int dstChannels, int srcStartChan, int sr
/*
if(!usedirectbuf)
{
- if(config.useDenormalBias)
+ if(MusEConfig::config.useDenormalBias)
{
for(unsigned int q = 0; q < nframes; ++q)
- outBuffers[i][q] = denormalBias;
+ outBuffers[i][q] = MusEGlobal::denormalBias;
}
else
memset(outBuffers[i], 0, sizeof(float) * nframes);
@@ -847,14 +862,14 @@ void AudioTrack::addData(unsigned pos, int dstChannels, int srcStartChan, int sr
unsigned int q;
for(i = 0; i < srcChans; ++i)
{
- if(config.useDenormalBias)
+ if(MusEConfig::config.useDenormalBias)
{
for(q = 0; q < nframes; ++q)
{
if(q & 1)
- buffer[i][q] -= denormalBias;
+ buffer[i][q] -= MusEGlobal::denormalBias;
else
- buffer[i][q] += denormalBias;
+ buffer[i][q] += MusEGlobal::denormalBias;
}
}
}
@@ -943,10 +958,10 @@ void AudioTrack::addData(unsigned pos, int dstChannels, int srcStartChan, int sr
{
for(i = 0; i < srcChannels; ++i)
{
- if(config.useDenormalBias)
+ if(MusEConfig::config.useDenormalBias)
{
for(unsigned int q = 0; q < nframes; ++q)
- outBuffers[i][q] = denormalBias;
+ outBuffers[i][q] = MusEGlobal::denormalBias;
}
else
memset(outBuffers[i], 0, sizeof(float) * nframes);
@@ -979,10 +994,10 @@ void AudioTrack::addData(unsigned pos, int dstChannels, int srcStartChan, int sr
unsigned int q;
for(i = 0; i < dstChannels; ++i)
{
- if(config.useDenormalBias)
+ if(MusEConfig::config.useDenormalBias)
{
for(q = 0; q < nframes; q++)
- dstBuffer[i][q] = denormalBias;
+ dstBuffer[i][q] = MusEGlobal::denormalBias;
}
else
memset(dstBuffer[i], 0, sizeof(float) * nframes);
@@ -1313,7 +1328,7 @@ bool AudioTrack::getData(unsigned pos, int channels, unsigned nframes, float** b
bool AudioInput::getData(unsigned, int channels, unsigned nframes, float** buffer)
{
- if (!checkAudioDevice()) return false;
+ if (!MusEGlobal::checkAudioDevice()) return false;
for (int ch = 0; ch < channels; ++ch)
{
void* jackPort = jackPorts[ch];
@@ -1337,10 +1352,10 @@ bool AudioInput::getData(unsigned, int channels, unsigned nframes, float** buffe
//memcpy(buffer[ch], jackbuf, nframes* sizeof(float));
AL::dsp->cpy(buffer[ch], jackbuf, nframes);
- if (config.useDenormalBias)
+ if (MusEConfig::config.useDenormalBias)
{
for (unsigned int i=0; i < nframes; i++)
- buffer[ch][i] += denormalBias;
+ buffer[ch][i] += MusEGlobal::denormalBias;
// p3.3.41
//fprintf(stderr, "AudioInput::getData %s Jack port %p efx apply channels:%d nframes:%ld %e %e %e %e\n",
@@ -1349,10 +1364,10 @@ bool AudioInput::getData(unsigned, int channels, unsigned nframes, float** buffe
}
else
{
- if (config.useDenormalBias)
+ if (MusEConfig::config.useDenormalBias)
{
for (unsigned int i=0; i < nframes; i++)
- buffer[ch][i] = denormalBias;
+ buffer[ch][i] = MusEGlobal::denormalBias;
}
else
{
@@ -1374,7 +1389,7 @@ bool AudioInput::getData(unsigned, int channels, unsigned nframes, float** buffe
void AudioInput::setName(const QString& s)
{
_name = s;
- if (!checkAudioDevice()) return;
+ if (!MusEGlobal::checkAudioDevice()) return;
for (int i = 0; i < channels(); ++i) {
char buffer[128];
snprintf(buffer, 128, "%s-%d", _name.toLatin1().constData(), i);
@@ -1490,7 +1505,7 @@ void AudioTrack::record()
while(fifo.getCount()) {
- if (fifo.get(_channels, segmentSize, buffer, &pos)) {
+ if (fifo.get(_channels, MusEGlobal::segmentSize, buffer, &pos)) {
printf("AudioTrack::record(): empty fifo\n");
return;
}
@@ -1546,7 +1561,7 @@ void AudioTrack::record()
//printf("AudioTrack::record loopcnt:%d lframe:%d newpos:%d curpos:%d start:%d end:%d\n", audio->loopCount(), audio->loopFrame(), pos, position, audio->getStartRecordPos().frame(), audio->getEndRecordPos().frame());
_recFile->seek(pos, 0);
- _recFile->write(_channels, buffer, segmentSize);
+ _recFile->write(_channels, buffer, MusEGlobal::segmentSize);
}
}
@@ -1563,13 +1578,13 @@ void AudioTrack::record()
void AudioOutput::processInit(unsigned nframes)
{
_nframes = nframes;
- if (!checkAudioDevice()) return;
+ if (!MusEGlobal::checkAudioDevice()) return;
for (int i = 0; i < channels(); ++i) {
if (jackPorts[i]) {
buffer[i] = audioDevice->getBuffer(jackPorts[i], nframes);
- if (config.useDenormalBias) {
+ if (MusEConfig::config.useDenormalBias) {
for (unsigned int j=0; j < nframes; j++)
- buffer[i][j] += denormalBias;
+ buffer[i][j] += MusEGlobal::denormalBias;
}
}
else
@@ -1607,9 +1622,9 @@ void AudioOutput::silence(unsigned n)
{
processInit(n);
for (int i = 0; i < channels(); ++i)
- if (config.useDenormalBias) {
+ if (MusEConfig::config.useDenormalBias) {
for (unsigned int j=0; j < n; j++)
- buffer[i][j] = denormalBias;
+ buffer[i][j] = MusEGlobal::denormalBias;
} else {
memset(buffer[i], 0, n * sizeof(float));
}
@@ -1638,8 +1653,8 @@ void AudioOutput::processWrite()
}
}
// Changed by Tim. p3.3.18
- //if (audioClickFlag && song->click()) {
- if (sendMetronome() && audioClickFlag && song->click()) {
+ //if (MusEGlobal::audioClickFlag && song->click()) {
+ if (sendMetronome() && MusEGlobal::audioClickFlag && song->click()) {
// Added by Tim. p3.3.18
#ifdef METRONOME_DEBUG
@@ -1658,7 +1673,7 @@ void AudioOutput::processWrite()
void AudioOutput::setName(const QString& s)
{
_name = s;
- if (!checkAudioDevice()) return;
+ if (!MusEGlobal::checkAudioDevice()) return;
for (int i = 0; i < channels(); ++i) {
char buffer[128];
snprintf(buffer, 128, "%s-%d", _name.toLatin1().constData(), i);
@@ -1680,7 +1695,7 @@ Fifo::Fifo()
{
muse_atomic_init(&count);
//nbuffer = FIFO_BUFFER;
- nbuffer = fifoLength;
+ nbuffer = MusEGlobal::fifoLength;
buffer = new FifoBuffer*[nbuffer];
for (int i = 0; i < nbuffer; ++i)
buffer[i] = new FifoBuffer;
@@ -1915,7 +1930,7 @@ void AudioTrack::setTotalOutChannels(int num)
outBuffers = new float*[chans];
for (int i = 0; i < chans; ++i)
- posix_memalign((void**)&outBuffers[i], 16, sizeof(float) * segmentSize);
+ 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.
generated by cgit v1.2.3 (git 2.39.1) at 2024-08-30 20:59:53 +0000