diff options
Diffstat (limited to 'muse2/muse/node.cpp')
| -rw-r--r-- | muse2/muse/node.cpp | 769 |
1 files changed, 344 insertions, 425 deletions
diff --git a/muse2/muse/node.cpp b/muse2/muse/node.cpp index dd41ba10..cb77f939 100644 --- a/muse2/muse/node.cpp +++ b/muse2/muse/node.cpp @@ -4,6 +4,7 @@ // $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) +// (C) Copyright 2011 Tim E. Real (terminator356 on sourceforge) // // This program is free software; you can redistribute it and/or // modify it under the terms of the GNU General Public License @@ -25,6 +26,8 @@ #include <sndfile.h> #include <stdlib.h> +#include <QString> + #include "node.h" #include "globals.h" #include "gconfig.h" @@ -44,7 +47,6 @@ //#define NODE_DEBUG //#define FIFO_DEBUG -// Added by Tim. p3.3.18 //#define METRONOME_DEBUG namespace MusECore { @@ -162,10 +164,19 @@ void Track::updateInternalSoloStates() void MidiTrack::updateInternalSoloStates() { - if(this == _tmpSoloChainTrack) - return; - - Track::updateInternalSoloStates(); + if(_nodeTraversed) // Anti circular mechanism. + { + fprintf(stderr, "MidiTrack::updateInternalSoloStates %s :\n MusE Warning: Please check your routes: Circular path found!\n", name().toLatin1().constData()); + return; + } + //if(this == _tmpSoloChainTrack) + // return; + + _nodeTraversed = true; + + Track::updateInternalSoloStates(); + + _nodeTraversed = false; // Reset. } @@ -175,40 +186,61 @@ void MidiTrack::updateInternalSoloStates() void AudioTrack::updateInternalSoloStates() { - if(this == _tmpSoloChainTrack) - return; - - Track::updateInternalSoloStates(); - - if(_tmpSoloChainDoIns) + if(_nodeTraversed) // Anti circular mechanism. + { + fprintf(stderr, "AudioTrack::updateInternalSoloStates %s :\n MusE Warning: Please check your routes: Circular path found!\n", name().toLatin1().constData()); + return; + } + //if(this == _tmpSoloChainTrack) + // return; + + _nodeTraversed = true; + + Track::updateInternalSoloStates(); + + if(_tmpSoloChainDoIns) + { + if(type() == AUDIO_SOFTSYNTH) + { + const MusECore::MidiTrackList* ml = MusEGlobal::song->midis(); + for(MusECore::ciMidiTrack im = ml->begin(); im != ml->end(); ++im) { - if(type() == AUDIO_SOFTSYNTH) - { - const MusECore::MidiTrackList* ml = MusEGlobal::song->midis(); - for(MusECore::ciMidiTrack im = ml->begin(); im != ml->end(); ++im) - { - MusECore::MidiTrack* mt = *im; - if(mt->outPort() >= 0 && mt->outPort() == ((SynthI*)this)->midiPort()) - mt->updateInternalSoloStates(); - } - } - - const RouteList* rl = inRoutes(); - for(ciRoute ir = rl->begin(); ir != rl->end(); ++ir) - { - if(ir->type == Route::TRACK_ROUTE) - ir->track->updateInternalSoloStates(); - } + MusECore::MidiTrack* mt = *im; + if(mt->outPort() >= 0 && mt->outPort() == ((SynthI*)this)->midiPort()) + mt->updateInternalSoloStates(); } - else - { - const RouteList* rl = outRoutes(); - for(ciRoute ir = rl->begin(); ir != rl->end(); ++ir) + } + + const RouteList* rl = inRoutes(); + for(ciRoute ir = rl->begin(); ir != rl->end(); ++ir) + { + if(ir->type == Route::TRACK_ROUTE) + ir->track->updateInternalSoloStates(); + else + // Support Midi Port -> Audio Input solo chains. p4.0.37 Tim. + if(ir->type == Route::MIDI_PORT_ROUTE) + { + const MidiTrackList* ml = MusEGlobal::song->midis(); + for(ciMidiTrack im = ml->begin(); im != ml->end(); ++im) { - if(ir->type == Route::TRACK_ROUTE) - ir->track->updateInternalSoloStates(); + MidiTrack* mt = *im; + if(mt->outPort() == ir->midiPort && ((1 << mt->outChannel()) & ir->channel) ) + mt->updateInternalSoloStates(); } - } + } + } + } + else + { + const RouteList* rl = outRoutes(); + for(ciRoute ir = rl->begin(); ir != rl->end(); ++ir) + { + if(ir->type == Route::TRACK_ROUTE) + ir->track->updateInternalSoloStates(); + } + } + + _nodeTraversed = false; // Reset. } @@ -221,6 +253,8 @@ void MidiTrack::updateSoloStates(bool noDec) if(noDec && !_solo) return; + _nodeTraversed = true; // Anti circular mechanism. + _tmpSoloChainTrack = this; _tmpSoloChainDoIns = false; _tmpSoloChainNoDec = noDec; @@ -244,6 +278,8 @@ void MidiTrack::updateSoloStates(bool noDec) } } } + + _nodeTraversed = false; // Reset. } @@ -256,6 +292,8 @@ void AudioTrack::updateSoloStates(bool noDec) if(noDec && !_solo) return; + _nodeTraversed = true; // Anti circular mechanism. + _tmpSoloChainTrack = this; _tmpSoloChainNoDec = noDec; updateSoloState(); @@ -301,6 +339,8 @@ void AudioTrack::updateSoloStates(bool noDec) ir->track->updateInternalSoloStates(); } } + + _nodeTraversed = false; // Reset. } @@ -327,7 +367,6 @@ void Track::setOff(bool val) // copyData //--------------------------------------------------------- -//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. @@ -336,60 +375,51 @@ void AudioTrack::copyData(unsigned pos, int dstChannels, int srcStartChan, int s // 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). + #ifdef NODE_DEBUG + printf("MusE: AudioTrack::copyData name:%s processed:%d\n", name().toLatin1().constData(), processed()); + #endif + if(srcStartChan == -1) srcStartChan = 0; - int srcChans = (srcChannels == -1) ? channels() : srcChannels; + int trackChans = channels(); + int srcChans = (srcChannels == -1) ? trackChans : srcChannels; int srcTotalOutChans = totalOutChannels(); if(channels() == 1) srcTotalOutChans = 1; - #ifdef NODE_DEBUG - printf("MusE: AudioTrack::copyData name:%s processed:%d\n", name().toLatin1().constData(), processed()); - #endif - // 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 ! // 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); - bool usedirectbuf = ((outRoutes()->size() <= 1) || (type() == AUDIO_OUTPUT)) && (this != metronome); + //bool usedirectbuf = ((outRoutes()->size() <= 1) || (type() == AUDIO_OUTPUT)) && (this != metronome); int i; - // p3.3.38 - //float* buffer[srcChannels]; float* buffer[srcTotalOutChans]; + float data[nframes * srcTotalOutChans]; - - //float data[nframes * srcChannels]; - //for(i = 0; i < srcChannels; ++i) - // buffer[i] = data + i * nframes; - // precalculate stereo volume double vol[2]; double _volume = volume(); 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? if(processed()) { // If there is only one (or no) output routes, it's an error - we've been called more than once per process cycle! + // No, this is no longer an error, it's deliberate. Processing no longer done in 'chains', now done randomly. p4.0.37 #ifdef NODE_DEBUG - if(usedirectbuf) - printf("MusE: AudioTrack::copyData Error! One or no out routes, but already processed! Copying local buffers anyway...\n"); + printf("MusE: AudioTrack::copyData name:%s already processed _haveData:%d\n", name().toLatin1().constData(), _haveData); #endif // Is there already some data gathered from a previous call during this process cycle? 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) - // buffer[i] = outBuffers[i]; - // p3.3.38 for(i = 0; i < srcTotalOutChans; ++i) buffer[i] = outBuffers[i]; } @@ -413,29 +443,16 @@ void AudioTrack::copyData(unsigned pos, int dstChannels, int srcStartChan, int s { // 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; - // 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 in AudioInput::getData! - // 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)) - // p3.3.38 - if(off() || !getData(pos, srcTotalOutChans, nframes, buffer) || (isMute() && !_prefader)) - { + _haveData = false; // Reset. + _processed = true; // Set this now. + + if(off()) + { #ifdef NODE_DEBUG - printf("MusE: AudioTrack::copyData name:%s dstChannels:%d zeroing buffers\n", name().toLatin1().constData(), dstChannels); + printf("MusE: AudioTrack::copyData name:%s dstChannels:%d Off, zeroing buffers\n", name().toLatin1().constData(), dstChannels); #endif - // No data was available. Zero the supplied buffers. + // Track is off. Zero the supplied buffers. unsigned int q; for(i = 0; i < dstChannels; ++i) { @@ -448,13 +465,12 @@ void AudioTrack::copyData(unsigned pos, int dstChannels, int srcStartChan, int s memset(dstBuffer[i], 0, sizeof(float) * nframes); } - for(i = 0; i < srcChans; ++i) + _efxPipe->apply(0, nframes, 0); // Just process controls only, not audio (do not 'run'). + + for(i = 0; i < trackChans; ++i) { - //_meter[i] = 0; _meter[i] = 0.0; - - /* - if(!usedirectbuf) + /*if(!usedirectbuf) { if(MusEGlobal::config.useDenormalBias) { @@ -463,22 +479,49 @@ void AudioTrack::copyData(unsigned pos, int dstChannels, int srcStartChan, int s } else memset(outBuffers[i], 0, sizeof(float) * nframes); - } - */ + } */ } - _haveData = false; - _processed = true; + //_haveData = false; + //_processed = true; + //_isProcessing = false; // Unblock. return; } + + // Point the input buffers at a temporary stack buffer. + 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 in AudioInput::getData! + // 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, srcTotalOutChans, nframes, buffer) || (isMute() && !_prefader)) + { + #ifdef NODE_DEBUG + printf("MusE: AudioTrack::copyData name:%s srcTotalOutChans:%d zeroing buffers\n", name().toLatin1().constData(), srcTotalOutChans); + #endif + + // No data was available. Track is not off. Zero the working buffers and continue on. + unsigned int q; + for(i = 0; i < srcTotalOutChans; ++i) + { + if(MusEGlobal::config.useDenormalBias) + { + for(q = 0; q < nframes; ++q) + buffer[i][q] = MusEGlobal::denormalBias; + } + else + memset(buffer[i], 0, sizeof(float) * nframes); + } + } //--------------------------------------------------- // apply plugin chain //--------------------------------------------------- - // p3.3.41 //fprintf(stderr, "AudioTrack::copyData %s efx apply srcChans:%d\n", name().toLatin1().constData(), srcChans); - _efxPipe->apply(srcChans, nframes, buffer); + _efxPipe->apply(trackChans, nframes, buffer); //--------------------------------------------------- // aux sends @@ -525,18 +568,16 @@ void AudioTrack::copyData(unsigned pos, int dstChannels, int srcStartChan, int s if(_prefader) { - for(i = 0; i < srcChans; ++i) + for(i = 0; i < trackChans; ++i) { float* p = buffer[i]; meter[i] = 0.0; for(unsigned k = 0; k < nframes; ++k) { - double f = fabs(*p); + double f = fabs(*p++); if(f > meter[i]) meter[i] = f; - ++p; } - //_meter[i] = lrint(meter[i] * 32767.0); _meter[i] = meter[i]; if(_meter[i] > _peak[i]) _peak[i] = _meter[i]; @@ -570,20 +611,19 @@ void AudioTrack::copyData(unsigned pos, int dstChannels, int srcStartChan, int s else memset(outBuffers[i], 0, sizeof(float) * nframes); } - } - */ + } */ + - _haveData = false; - _processed = true; + if(!_prefader) + for(i = 0; i < trackChans; ++i) // Must process ALL channels, even if unconnected. Only max 2 channels. + _meter[i] = 0.0; + return; } // 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) + //if(!usedirectbuf) { - //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); } @@ -606,9 +646,9 @@ void AudioTrack::copyData(unsigned pos, int dstChannels, int srcStartChan, int s 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; @@ -618,132 +658,94 @@ void AudioTrack::copyData(unsigned pos, int dstChannels, int srcStartChan, int s // postfader metering //--------------------------------------------------- - - if(srcChans == dstChannels) + #ifdef NODE_DEBUG + printf("MusE: AudioTrack::copyData trackChans:%d srcTotalOutChans:%d srcStartChan:%d srcChans:%d dstChannels:%d\n", trackChans, srcTotalOutChans, srcStartChan, srcChans, dstChannels); + #endif + + if(!_prefader) { - if(_prefader) + for(int c = 0; c < trackChans; ++c) { - for(int c = 0; c < dstChannels; ++c) + meter[c] = 0.0; + double v = (trackChans == 1 ? _volume : vol[c]); + float* sp = buffer[c]; + for(unsigned k = 0; k < nframes; ++k) { - // 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]); + float val = *sp++ * v; // If the track is mono pan has no effect on meters. + double f = fabs(val); + if(f > meter[c]) + meter[c] = f; } - } - else + _meter[c] = meter[c]; + if(_meter[c] > _peak[c]) + _peak[c] = _meter[c]; + } + } + + if(srcChans == dstChannels) + { + for(int c = 0; c < dstChannels; ++c) { - for(int c = 0; c < dstChannels; ++c) - { - meter[c] = 0.0; - - // 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) - { - float val = *sp++ * vol[c]; - *dp++ = val; - double f = fabs(val); - if(f > meter[c]) - meter[c] = f; - } - //_meter[c] = lrint(meter[c] * 32767.0); - _meter[c] = meter[c]; - if(_meter[c] > _peak[c]) - _peak[c] = _meter[c]; - } + double v; + if(srcStartChan > 2) // Don't apply pan or volume to extra channels above 2. + //v = _volume; + v = 1.0; + else + if(srcChans >= 2) // If 2 channels apply pan normally. + v = vol[c]; + else + if(trackChans < 2) // If 1 channel and track is 1 channel, don't apply pan. + v = _volume; + else + v = vol[srcStartChan]; // Otherwise 1 channel but track is 2 channels. Apply the channel volume. + + float* sp = buffer[c + srcStartChan]; + float* dp = dstBuffer[c]; + for(unsigned k = 0; k < nframes; ++k) + //*dp++ = (*sp++ * vol[c]); + *dp++ = (*sp++ * v); } } else if(srcChans == 1 && dstChannels == 2) { - // p3.3.38 - //float* sp = buffer[0]; - float* sp = buffer[srcStartChan]; - - if(_prefader) + for(int c = 0; c < dstChannels; ++c) { - for(int c = 0; c < dstChannels; ++c) - { - float* dp = dstBuffer[c]; - for(unsigned k = 0; k < nframes; ++k) - *dp++ = (*sp++ * vol[c]); - } - } - else - { - meter[0] = 0.0; - for(unsigned k = 0; k < nframes; ++k) - { - float val = *sp++; - double f = fabs(val) * _volume; - if(f > meter[0]) - meter[0] = f; - *(dstBuffer[0] + k) = val * vol[0]; - *(dstBuffer[1] + k) = val * vol[1]; - } - //_meter[0] = lrint(meter[0] * 32767.0); - _meter[0] = meter[0]; - if(_meter[0] > _peak[0]) - _peak[0] = _meter[0]; + double v; + if(srcStartChan > 2) // Don't apply pan or volume to extra channels above 2. + //v = _volume; + v = 1.0; + else + if(trackChans <= 1) // If track is mono apply pan. + v = vol[c]; + else + v = vol[srcStartChan]; // Otherwise track is stereo, apply the same channel volume to both. + + 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) { - // p3.3.38 - //float* sp1 = buffer[0]; - //float* sp2 = buffer[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 ? 1.0 : vol[srcStartChan]); // Don't apply pan or volume to extra channels above 2. + double v2 = (srcStartChan > 2 ? 1.0 : vol[srcStartChan + 1]); // + float* dp = dstBuffer[0]; float* sp1 = buffer[srcStartChan]; float* sp2 = buffer[srcStartChan + 1]; - - if(_prefader) - { - float* dp = dstBuffer[0]; - for(unsigned k = 0; k < nframes; ++k) - *dp++ = (*sp1++ * vol[0] + *sp2++ * vol[1]); - } - else - { - float* dp = dstBuffer[0]; - meter[0] = 0.0; - meter[1] = 0.0; - for(unsigned k = 0; k < nframes; ++k) - { - float val1 = *sp1++ * vol[0]; - float val2 = *sp2++ * vol[1]; - double f1 = fabs(val1); - if(f1 > meter[0]) - meter[0] = f1; - double f2 = fabs(val2); - if(f2 > meter[1]) - meter[1] = f2; - *dp++ = (val1 + val2); - } - //_meter[0] = lrint(meter[0] * 32767.0); - _meter[0] = meter[0]; - if(_meter[0] > _peak[0]) - _peak[0] = _meter[0]; - //_meter[1] = lrint(meter[1] * 32767.0); - _meter[1] = meter[1]; - if(_meter[1] > _peak[1]) - _peak[1] = _meter[1]; - } + for(unsigned k = 0; k < nframes; ++k) + //*dp++ = (*sp1++ * vol[0] + *sp2++ * vol[1]); + *dp++ = (*sp1++ * v1 + *sp2++ * v2); } - - _processed = true; } //--------------------------------------------------------- // addData //--------------------------------------------------------- -//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. @@ -752,21 +754,21 @@ void AudioTrack::addData(unsigned pos, int dstChannels, int srcStartChan, int sr // 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). - //Added by Tim. p3.3.16 #ifdef NODE_DEBUG printf("MusE: AudioTrack::addData name:%s processed:%d\n", name().toLatin1().constData(), processed()); #endif - if (off()) - { - _processed = true; - return; - } + //if (off()) + //{ + // _processed = true; + // return; + //} if(srcStartChan == -1) srcStartChan = 0; - int srcChans = (srcChannels == -1) ? channels() : srcChannels; + int trackChans = channels(); + int srcChans = (srcChannels == -1) ? trackChans : srcChannels; int srcTotalOutChans = totalOutChannels(); if(channels() == 1) srcTotalOutChans = 1; @@ -774,18 +776,12 @@ void AudioTrack::addData(unsigned pos, int dstChannels, int srcStartChan, int sr // 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 ! // 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); - bool usedirectbuf = ((outRoutes()->size() <= 1) || (type() == AUDIO_OUTPUT)) && (this != metronome); + //bool usedirectbuf = ((outRoutes()->size() <= 1) || (type() == AUDIO_OUTPUT)) && (this != metronome); int i; - // p3.3.38 - //float* buffer[srcChannels]; float* buffer[srcTotalOutChans]; - - //float data[nframes * srcChannels]; - //for (i = 0; i < srcChannels; ++i) - // buffer[i] = data + i * nframes; + float data[nframes * srcTotalOutChans]; // precalculate stereo volume double vol[2]; @@ -793,105 +789,96 @@ 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[srcChans]; + float meter[trackChans]; // Have we been here already during this process cycle? if(processed()) { // If there is only one (or no) output routes, it's an error - we've been called more than once per process cycle! + // No, this is no longer an error, it's deliberate. Processing no longer done in 'chains', now done randomly. p4.0.37 #ifdef NODE_DEBUG - if(usedirectbuf) - printf("MusE: AudioTrack::addData Error! One or no out routes, but already processed! Copying local buffers anyway...\n"); + printf("MusE: AudioTrack::addData name:%s already processed _haveData:%d\n", name().toLatin1().constData(), _haveData); #endif // Is there already some data gathered from a previous call during this process cycle? 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) - // buffer[i] = outBuffers[i]; - // p3.3.38 for(i = 0; i < srcTotalOutChans; ++i) buffer[i] = outBuffers[i]; } else + { // No data was available from a previous call during this process cycle. Nothing to add, just return. return; + } } else { // 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; - // p3.3.38 - float data[nframes * srcTotalOutChans]; - for(i = 0; i < srcTotalOutChans; ++i) - buffer[i] = data + i * nframes; - + _haveData = false; // Reset. + _processed = true; // Set this now. - // 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)) - // 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 < srcChans; ++i) + if(off()) + { + #ifdef NODE_DEBUG + printf("MusE: AudioTrack::addData name:%s dstChannels:%d Track is Off \n", name().toLatin1().constData(), dstChannels); + #endif + + // Nothing to zero or add... + + _efxPipe->apply(0, nframes, 0); // Track is off. Just process controls only, not audio (do not 'run'). + + for(i = 0; i < trackChans; ++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. - /* - if(!usedirectbuf) + _meter[i] = 0.0; + /*if(!usedirectbuf) { if(MusEGlobal::config.useDenormalBias) { - for(unsigned int q = 0; q < nframes; ++q) + for(q = 0; q < nframes; ++q) outBuffers[i][q] = MusEGlobal::denormalBias; - } - else + } + else memset(outBuffers[i], 0, sizeof(float) * nframes); - } - */ - - //_meter[i] = 0; - _meter[i] = 0.0; - } - - _haveData = false; - _processed = true; + } */ + } return; } - - /* - // p3.3.41 Added. - unsigned int q; - for(i = 0; i < srcChans; ++i) + + // Point the input buffers at a temporary stack buffer. + 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, srcTotalOutChans, nframes, buffer)) { - if(MusEGlobal::config.useDenormalBias) - { - for(q = 0; q < nframes; ++q) + // No data was available. Track is not off. Zero the working buffers and continue on. + unsigned int q; + for(i = 0; i < srcTotalOutChans; ++i) + { + if(MusEGlobal::config.useDenormalBias) { - if(q & 1) - buffer[i][q] -= MusEGlobal::denormalBias; - else - buffer[i][q] += MusEGlobal::denormalBias; - } - } - } - */ - + for(q = 0; q < nframes; ++q) + buffer[i][q] = MusEGlobal::denormalBias; + } + else + memset(buffer[i], 0, sizeof(float) * nframes); + } + } + //--------------------------------------------------- // apply plugin chain //--------------------------------------------------- - // p3.3.41 //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(srcChans, nframes, buffer); - // p3.3.41 + + _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]); @@ -940,18 +927,16 @@ void AudioTrack::addData(unsigned pos, int dstChannels, int srcStartChan, int sr if(_prefader) { - for(i = 0; i < srcChans; ++i) + for(i = 0; i < trackChans; ++i) { float* p = buffer[i]; meter[i] = 0.0; for(unsigned k = 0; k < nframes; ++k) { - double f = fabs(*p); + double f = fabs(*p++); if(f > meter[i]) meter[i] = f; - ++p; } - //_meter[i] = lrint(meter[i] * 32767.0); _meter[i] = meter[i]; if(_meter[i] > _peak[i]) _peak[i] = _meter[i]; @@ -961,8 +946,7 @@ 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) + /* if(!usedirectbuf) { for(i = 0; i < srcChannels; ++i) { @@ -974,20 +958,19 @@ void AudioTrack::addData(unsigned pos, int dstChannels, int srcStartChan, int sr else memset(outBuffers[i], 0, sizeof(float) * nframes); } - } - */ + } */ - _haveData = false; - _processed = true; + if(!_prefader) + //for(i = 0; i < srcChans; ++i) + for(i = 0; i < trackChans; ++i) + _meter[i] = 0.0; + return; } // 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) + //if(!usedirectbuf) { - //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); } @@ -1010,9 +993,9 @@ void AudioTrack::addData(unsigned pos, int dstChannels, int srcStartChan, int sr 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; @@ -1022,123 +1005,88 @@ void AudioTrack::addData(unsigned pos, int dstChannels, int srcStartChan, int sr // postfader metering //--------------------------------------------------- - if(srcChans == dstChannels) + #ifdef NODE_DEBUG + printf("MusE: AudioTrack::addData trackChans:%d srcTotalOutChans:%d srcChans:%d dstChannels:%d\n", trackChans, srcTotalOutChans, srcChans, dstChannels); + #endif + + if(!_prefader) { - if(_prefader) + for(int c = 0; c < trackChans; ++c) { - for(int c = 0; c < dstChannels; ++c) + meter[c] = 0.0; + double v = (trackChans == 1 ? _volume : vol[c]); + float* sp = buffer[c]; + for(unsigned k = 0; k < nframes; ++k) { - // 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]); + float val = *sp++ * v; // If the track is mono pan has no effect on meters. + double f = fabs(val); + if(f > meter[c]) + meter[c] = f; } - } - else + _meter[c] = meter[c]; + if(_meter[c] > _peak[c]) + _peak[c] = _meter[c]; + } + } + + if(srcChans == dstChannels) + { + for(int c = 0; c < dstChannels; ++c) { - for(int c = 0; c < dstChannels; ++c) - { - meter[c] = 0.0; - // p3.3.38 - //float* sp = buffer[c]; - float* sp = buffer[c + srcStartChan]; - - float* dp = dstBuffer[c]; - for(unsigned k = 0; k < nframes; ++k) - { - float val = *sp++ * vol[c]; - *dp++ += val; - double f = fabs(val); - if (f > meter[c]) - meter[c] = f; - } - //_meter[c] = lrint(meter[c] * 32767.0); - _meter[c] = meter[c]; - if(_meter[c] > _peak[c]) - _peak[c] = _meter[c]; - } + double v; + if(srcStartChan > 2) // Don't apply pan or volume to extra channels above 2. + //v = _volume; + v = 1.0; + else + if(srcChans >= 2) // If 2 channels apply pan normally. + v = vol[c]; + else + if(trackChans < 2) // If 1 channel and track is 1 channel, don't apply pan. + v = _volume; + else + v = vol[srcStartChan]; // Otherwise 1 channel but track is 2 channels. Apply the channel volume. + + float* sp = buffer[c + srcStartChan]; + float* dp = dstBuffer[c]; + for(unsigned k = 0; k < nframes; ++k) + //*dp++ += (*sp++ * vol[c]); + *dp++ += (*sp++ * v); } } else if(srcChans == 1 && dstChannels == 2) { - // p3.3.38 - float* sp = buffer[srcStartChan]; - - if(_prefader) + for(int c = 0; c < dstChannels; ++c) { - for(int c = 0; c < dstChannels; ++c) - { - float* dp = dstBuffer[c]; - //float* sp = buffer[0]; - for(unsigned k = 0; k < nframes; ++k) - *dp++ += (*sp++ * vol[c]); - } - } - else - { - //float* sp = buffer[0]; - meter[0] = 0.0; - for(unsigned k = 0; k < nframes; ++k) - { - float val = *sp++; - double f = fabs(val) * _volume; - if(f > meter[0]) - meter[0] = f; - *(dstBuffer[0] + k) += val * vol[0]; - *(dstBuffer[1] + k) += val * vol[1]; - } - //_meter[0] = lrint(meter[0] * 32767.0); - _meter[0] = meter[0]; - if(_meter[0] > _peak[0]) - _peak[0] = _meter[0]; + double v; + if(srcStartChan > 2) // Don't apply pan or volume to extra channels above 2. + //v = _volume; + v = 1.0; + else + if(trackChans <= 1) // If track is mono apply pan. + v = vol[c]; + else + v = vol[srcStartChan]; // Otherwise track is stereo, apply the same channel volume to both. + + 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) { - // p3.3.38 - //float* sp1 = buffer[0]; - //float* sp2 = buffer[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 ? 1.0 : vol[srcStartChan]); // Don't apply pan or volume to extra channels above 2. + double v2 = (srcStartChan > 2 ? 1.0 : vol[srcStartChan + 1]); // float* sp1 = buffer[srcStartChan]; float* sp2 = buffer[srcStartChan + 1]; - - if(_prefader) - { - float* dp = dstBuffer[0]; - for(unsigned k = 0; k < nframes; ++k) - *dp++ += (*sp1++ * vol[0] + *sp2++ * vol[1]); - } - else - { - float* dp = dstBuffer[0]; - meter[0] = 0.0; - meter[1] = 0.0; - for(unsigned k = 0; k < nframes; ++k) - { - float val1 = *sp1++ * vol[0]; - float val2 = *sp2++ * vol[1]; - double f1 = fabs(val1); - if(f1 > meter[0]) - meter[0] = f1; - double f2 = fabs(val2); - if(f2 > meter[1]) - meter[1] = f2; - *dp++ += (val1 + val2); - } - //_meter[0] = lrint(meter[0] * 32767.0); - _meter[0] = meter[0]; - if(_meter[0] > _peak[0]) - _peak[0] = _meter[0]; - //_meter[1] = lrint(meter[1] * 32767.0); - _meter[1] = meter[1]; - if(_meter[1] > _peak[1]) - _peak[1] = _meter[1]; - } + float* dp = dstBuffer[0]; + for(unsigned k = 0; k < nframes; ++k) + //*dp++ += (*sp1++ * vol[0] + *sp2++ * vol[1]); + *dp++ += (*sp1++ * v1 + *sp2++ * v2); } - - _processed = true; } //--------------------------------------------------------- @@ -1300,15 +1248,12 @@ bool AudioTrack::getData(unsigned pos, int channels, unsigned nframes, float** b printf(" calling copyData on %s...\n", ir->track->name().toLatin1().constData()); #endif - // 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); - // p3.3.41 //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; @@ -1320,8 +1265,6 @@ bool AudioTrack::getData(unsigned pos, int channels, unsigned nframes, float** b 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, @@ -1344,12 +1287,11 @@ bool AudioInput::getData(unsigned, int channels, unsigned nframes, float** buffe void* jackPort = jackPorts[ch]; //float* jackbuf = 0; - //if (jackPort) { - // p3.3.41 Do not get buffers of unconnected client ports. Causes repeating leftover data, can be loud, or DC ! + // Do not get buffers of unconnected client ports. Causes repeating leftover data, can be loud, or DC ! if (jackPort && MusEGlobal::audioDevice->connections(jackPort)) { //buffer[ch] = MusEGlobal::audioDevice->getBuffer(jackPort, nframes); - // p3.3.41 If the client port buffer is also used by another channel (connected to the same jack port), + // If the client port buffer is also used by another channel (connected to the same jack port), // don't directly set pointer, copy the data instead. // Otherwise the next channel will interfere - it will overwrite the buffer ! // Verified symptoms: Can't use a splitter. Mono noise source on a stereo track sounds in mono. Etc... @@ -1366,8 +1308,6 @@ bool AudioInput::getData(unsigned, int channels, unsigned nframes, float** buffe { for (unsigned int i=0; i < nframes; i++) 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", // name().toLatin1().constData(), jackPort, channels, nframes, buffer[0][0], buffer[0][1], buffer[0][2], buffer[0][3]); } @@ -1384,8 +1324,6 @@ bool AudioInput::getData(unsigned, int channels, unsigned nframes, float** buffe memset(buffer[ch], 0, nframes * sizeof(float)); } - // p3.3.41 - //fprintf(stderr, "AudioInput::getData %s No Jack port efx apply channels:%d nframes:%ld %e %e %e %e\n", // name().toLatin1().constData(), channels, nframes, buffer[0][0], buffer[0][1], buffer[0][2], buffer[0][3]); } } @@ -1524,7 +1462,7 @@ void AudioTrack::record() return; } if (_recFile) { - // Line removed by Tim. p3.3.8 Oct 28, 2009 + // Line removed by Tim. Oct 28, 2009 //_recFile->seek(pos, 0); // // Fix for recorded waves being shifted ahead by an amount @@ -1570,7 +1508,6 @@ void AudioTrack::record() if( (pos >= fr) && (!MusEGlobal::song->punchout() || (!MusEGlobal::song->loop() && pos < MusEGlobal::song->rPos().frame())) ) { pos -= fr; - // Added by Tim. p3.3.8 //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()); @@ -1614,7 +1551,6 @@ 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().toLatin1().constData(), processed()); #endif @@ -1622,9 +1558,6 @@ void AudioOutput::process(unsigned pos, unsigned offset, unsigned n) for (int i = 0; i < _channels; ++i) { buffer1[i] = buffer[i] + offset; } - - // p3.3.38 - //copyData(pos, _channels, n, buffer1); copyData(pos, _channels, -1, -1, n, buffer1); } @@ -1666,17 +1599,13 @@ void AudioOutput::processWrite() putFifo(_channels, _nframes, buffer); } } - // Changed by Tim. p3.3.18 + // Changed by Tim. //if (MusEGlobal::audioClickFlag && MusEGlobal::song->click()) { if (sendMetronome() && MusEGlobal::audioClickFlag && MusEGlobal::song->click()) { - // Added by Tim. p3.3.18 #ifdef METRONOME_DEBUG printf("MusE: AudioOutput::processWrite Calling metronome->addData frame:%u channels:%d frames:%lu\n", MusEGlobal::audio->pos().frame(), _channels, _nframes); #endif - - // p3.3.38 - //metronome->addData(MusEGlobal::audio->pos().frame(), _channels, _nframes, buffer); metronome->addData(MusEGlobal::audio->pos().frame(), _channels, -1, -1, _nframes, buffer); } } @@ -1721,7 +1650,6 @@ Fifo::~Fifo() { for (int i = 0; i < nbuffer; ++i) { - // p3.3.45 if(buffer[i]->buffer) { //printf("Fifo::~Fifo freeing buffer\n"); @@ -1742,7 +1670,6 @@ Fifo::~Fifo() bool Fifo::put(int segs, unsigned long samples, float** src, unsigned pos) { - // Added by Tim. p3.3.17 #ifdef FIFO_DEBUG printf("FIFO::put segs:%d samples:%lu pos:%u\n", segs, samples, pos); #endif @@ -1765,7 +1692,6 @@ bool Fifo::put(int segs, unsigned long samples, float** src, unsigned pos) // Changed by Tim. p3.3.15 //b->buffer = new float[n]; posix_memalign((void**)&(b->buffer), 16, sizeof(float) * n); - // p3.3.45 if(!b->buffer) { printf("Fifo::put could not allocate buffer segs:%d samples:%lu pos:%u\n", segs, samples, pos); @@ -1774,7 +1700,6 @@ bool Fifo::put(int segs, unsigned long samples, float** src, unsigned pos) b->maxSize = n; } - // p3.3.45 if(!b->buffer) { printf("Fifo::put no buffer! segs:%d samples:%lu pos:%u\n", segs, samples, pos); @@ -1798,7 +1723,6 @@ bool Fifo::put(int segs, unsigned long samples, float** src, unsigned pos) bool Fifo::get(int segs, unsigned long samples, float** dst, unsigned* pos) { - // Added by Tim. p3.3.17 #ifdef FIFO_DEBUG printf("FIFO::get segs:%d samples:%lu\n", segs, samples); #endif @@ -1808,7 +1732,6 @@ bool Fifo::get(int segs, unsigned long samples, float** dst, unsigned* pos) return true; } FifoBuffer* b = buffer[ridx]; - // p3.3.45 if(!b->buffer) { printf("Fifo::get no buffer! segs:%d samples:%lu b->pos:%u\n", segs, samples, b->pos); @@ -1844,7 +1767,6 @@ void Fifo::remove() bool Fifo::getWriteBuffer(int segs, unsigned long samples, float** buf, unsigned pos) { - // Added by Tim. p3.3.17 #ifdef FIFO_DEBUG printf("Fifo::getWriteBuffer segs:%d samples:%lu pos:%u\n", segs, samples, pos); #endif @@ -1866,7 +1788,6 @@ bool Fifo::getWriteBuffer(int segs, unsigned long samples, float** buf, unsigned // Changed by Tim. p3.3.15 //b->buffer = new float[n]; posix_memalign((void**)&(b->buffer), 16, sizeof(float) * n); - // p3.3.45 if(!b->buffer) { printf("Fifo::getWriteBuffer could not allocate buffer segs:%d samples:%lu pos:%u\n", segs, samples, pos); @@ -1875,8 +1796,6 @@ bool Fifo::getWriteBuffer(int segs, unsigned long samples, float** buf, unsigned b->maxSize = n; } - - // p3.3.45 if(!b->buffer) { printf("Fifo::getWriteBuffer no buffer! segs:%d samples:%lu pos:%u\n", segs, samples, pos); |