faust 2.79.3

Author: schollz

external/faust/architecture/faust/au/AUUI.h

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+/************************** BEGIN audio.h *****************************
+ FAUST Architecture File
+ Copyright (C) 2003-2022 GRAME, Centre National de Creation Musicale
+ ---------------------------------------------------------------------
+ This program is free software; you can redistribute it and/or modify
+ it under the terms of the GNU Lesser General Public License as published by
+ the Free Software Foundation; either version 2.1 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 Lesser General Public License for more details.
+ 
+ You should have received a copy of the GNU Lesser General Public License
+ along with this program; if not, write to the Free Software
+ Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA.
+ 
+ EXCEPTION : As a special exception, you may create a larger work
+ that contains this FAUST architecture section and distribute
+ that work under terms of your choice, so long as this FAUST
+ architecture section is not modified.
+ ***********************************************************************/
+
+#ifndef __audio__
+#define __audio__
+
+#include <set>
+#include <utility>
+
+class dsp;
+
+typedef void (* shutdown_callback)(const char* message, void* arg);
+
+typedef void (* compute_callback)(void* arg);
+
+class audio {
+    
+    protected:
+    
+        shutdown_callback fShutdown;    // Shutdown callback
+        void* fShutdownArg;             // Shutdown callback data
+    
+        std::set<std::pair<compute_callback, void*> > fComputeCallbackList;
+    
+    public:
+    
+        audio():fShutdown(nullptr), fShutdownArg(nullptr) {}
+        virtual ~audio() {}
+
+        /**
+         * Init the DSP.
+         * @param name - the DSP name to be given to the audio driven
+         * (could appear as a JACK client for instance)
+         * @param dsp - the dsp that will be initialized with the driver sample rate
+         *
+         * @return true is sucessful, false in case of driver failure.
+         **/
+        virtual bool init(const char* name, ::dsp* dsp) = 0;
+    
+        /**
+         * Start audio processing.
+         * @return true is sucessful, false if case of driver failure.
+         **/
+        virtual bool start() = 0;
+    
+        /**
+         * Stop audio processing.
+         **/
+        virtual void stop() = 0;
+    
+        void setShutdownCallback(shutdown_callback cb, void* arg)
+        {
+            fShutdown = cb;
+            fShutdownArg = arg;
+        }
+    
+        void addControlCallback(compute_callback cb, void* arg)
+        {
+            fComputeCallbackList.insert(std::make_pair(cb, arg));
+        }
+    
+        bool removeControlCallback(compute_callback cb, void* arg)
+        {
+            return (fComputeCallbackList.erase(std::make_pair(cb, arg)) == 1);
+        }
+    
+        void runControlCallbacks()
+        {
+            for (const auto& it : fComputeCallbackList) {
+                it.first(it.second);
+            }
+        }
+    
+        // Return buffer size in frames.
+        virtual int getBufferSize() = 0;
+    
+        // Return the driver sample rate in Hz.
+        virtual int getSampleRate() = 0;
+
+        // Return the driver hardware inputs number.
+        virtual int getNumInputs() = 0;
+    
+        // Return the driver hardware outputs number.
+        virtual int getNumOutputs() = 0;
+    
+        /**
+         * @return Returns the average proportion of available CPU
+         * being spent inside the audio callbacks (between 0.0 and 1.0).
+         **/
+        virtual float getCPULoad() { return 0.f; }
+};
+					
+#endif
+/**************************  END  audio.h **************************/

external/faust/architecture/faust/audio/alsa-dsp.h

@@ -0,0 +1,122 @@
+/************************** BEGIN channels.h **************************
+ FAUST Architecture File
+ Copyright (C) 2003-2022 GRAME, Centre National de Creation Musicale
+ ---------------------------------------------------------------------
+ This program is free software; you can redistribute it and/or modify
+ it under the terms of the GNU Lesser General Public License as published by
+ the Free Software Foundation; either version 2.1 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 Lesser General Public License for more details.
+ 
+ You should have received a copy of the GNU Lesser General Public License
+ along with this program; if not, write to the Free Software
+ Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA.
+ 
+ EXCEPTION : As a special exception, you may create a larger work
+ that contains this FAUST architecture section and distribute
+ that work under terms of your choice, so long as this FAUST
+ architecture section is not modified.
+ ************************************************************************/
+
+#ifndef __audio_channels__
+#define __audio_channels__
+
+#include <string.h>
+#include <assert.h>
+
+#ifndef FAUSTFLOAT
+#define FAUSTFLOAT float
+#endif
+
+template <class REAL>
+class real_channels
+{
+    private:
+        
+        int fNumFrames;
+        int fNumChannels;
+        REAL** fBuffers;
+        REAL** fSliceBuffers;
+        
+    public:
+        
+        real_channels(int nframes, int nchannels)
+        {
+            fBuffers = new REAL*[nchannels];
+            fSliceBuffers = new REAL*[nchannels];
+            fNumFrames = nframes;
+            fNumChannels = nchannels;
+            
+            // allocate audio channels
+            for (int chan = 0; chan < fNumChannels; chan++) {
+                fBuffers[chan] = new FAUSTFLOAT[fNumFrames];
+            }
+            
+            zero();
+        }
+        
+        void zero()
+        {
+            // clear audio channels
+            for (int chan = 0; chan < fNumChannels; chan++) {
+                memset(fBuffers[chan], 0, sizeof(REAL) * fNumFrames);
+            }
+        }
+        
+        void impulse()
+        {
+            // set first sample to 1 for all channels
+            for (int chan = 0; chan < fNumChannels; chan++) {
+                fBuffers[chan][0] = FAUSTFLOAT(1.0);
+                for (int frame = 1; frame < fNumFrames; frame++) {
+                    fBuffers[chan][frame] = REAL(0.0);
+                }
+            }
+        }
+        
+        void display()
+        {
+            for (int chan = 0; chan < fNumChannels; chan++) {
+                for (int frame = 0; frame < fNumFrames; frame++) {
+                    std::cout << "chan = " << chan << " frame = " << frame << " value = " << fBuffers[chan][frame] << std::endl;
+                }
+            }
+        }
+        
+        virtual ~real_channels()
+        {
+            // free separate input channels
+            for (int chan = 0; chan < fNumChannels; chan++) {
+                delete [] fBuffers[chan];
+            }
+            delete [] fBuffers;
+            delete [] fSliceBuffers;
+        }
+        
+        REAL** buffers() { return fBuffers; }
+        
+        REAL** buffers(int index)
+        {
+            assert(index < fNumFrames);
+            for (int chan = 0; chan < fNumChannels; chan++) {
+                fSliceBuffers[chan] = &fBuffers[chan][index];
+            }
+            return fSliceBuffers;
+        }
+    
+};
+
+class channels : public real_channels<FAUSTFLOAT> {
+
+    public:
+    
+        channels(int nframes, int nchannels):real_channels<FAUSTFLOAT>(nframes, nchannels) {}
+};
+
+#endif
+
+/**************************  END  channels.h **************************/