X-Git-Url: https://scm.cri.ensmp.fr/git/Faustine.git/blobdiff_plain/1059e1cc0c2ecfa237406949aa26155b6a5b9154..66f23d4fabf89ad09adbd4dfc15ac6b5b2b7da83:/interpreter/preprocessor/faust-0.9.47mr3/architecture/audio/portaudio-dsp.h

diff --git a/interpreter/preprocessor/faust-0.9.47mr3/architecture/audio/portaudio-dsp.h b/interpreter/preprocessor/faust-0.9.47mr3/architecture/audio/portaudio-dsp.h
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@@ -0,0 +1,192 @@
+/************************************************************************
+
+	IMPORTANT NOTE : this file contains two clearly delimited sections :
+	the ARCHITECTURE section (in two parts) and the USER section. Each section
+	is governed by its own copyright and license. Please check individually
+	each section for license and copyright information.
+*************************************************************************/
+
+/*******************BEGIN ARCHITECTURE SECTION (part 1/2)****************/
+
+/************************************************************************
+    FAUST Architecture File
+	Copyright (C) 2003-2011 GRAME, Centre National de Creation Musicale
+    ---------------------------------------------------------------------
+    This Architecture section 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; either version 3 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, see <http://www.gnu.org/licenses/>.
+
+	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 __portaudio_dsp__
+#define __portaudio_dsp__
+
+#include <stdio.h>
+#include <assert.h>
+#include <portaudio.h>
+
+#include "audio.h"
+#include "dsp.h"
+
+#define max(x,y) (((x)>(y)) ? (x) : (y))
+
+static int audioCallback(const void *ibuf, void *obuf, unsigned long frames, const PaStreamCallbackTimeInfo*,  PaStreamCallbackFlags, void * drv);
+//----------------------------------------------------------------------------
+static void pa_error(int err)
+{
+	if (err != paNoError) {
+		printf(  "PortAudio error: %s\n", Pa_GetErrorText( err ) );
+	}
+}
+
+/******************************************************************************
+*******************************************************************************
+
+							PORT AUDIO INTERFACE
+
+*******************************************************************************
+*******************************************************************************/
+class portaudio : public audio {
+
+        dsp*			fDsp;
+        PaStream*		fAudioStream;
+        long fSampleRate, fBufferSize;
+        //----------------------------------------------------------------------------
+        // 	number of physical input and output channels of the PA device
+        //----------------------------------------------------------------------------
+        int		fDevNumInChans;
+        int		fDevNumOutChans;
+
+        //----------------------------------------------------------------------------
+        // tables of noninterleaved input and output channels for FAUST
+        //----------------------------------------------------------------------------
+        float* 	fInChannel[256];
+        float* 	fOutChannel[256];
+
+        //----------------------------------------------------------------------------
+        // allocated the noninterleaved input and output channels for FAUST
+        //----------------------------------------------------------------------------
+        void allocChannels (int size, int numInChan, int numOutChan)
+        {
+            assert (numInChan < 256);
+            assert (numOutChan < 256);
+
+            for (int i = 0; i < numInChan; i++) {
+                fInChannel[i] = (float*) calloc (size, sizeof(float));
+                for (int j = 0; j < size; j++)
+                    fInChannel[i][j] = 0.0;
+            }
+
+            for (int i = 0; i < numOutChan; i++) {
+                fOutChannel[i] = (float*) calloc (size, sizeof(float));
+                for (int j = 0; j < size; j++)
+                    fOutChannel[i][j] = 0.0;
+            }
+        }
+
+	public:
+			 portaudio(long srate, long bsize) : fDsp(0), fAudioStream(0),
+				fSampleRate(srate), fBufferSize(bsize), fDevNumInChans(0), fDevNumOutChans(0) {}
+	virtual ~portaudio() { stop(); }
+
+	virtual bool init(const char* name, dsp* DSP){
+		fDsp = DSP;
+		pa_error(Pa_Initialize());
+
+		const PaDeviceInfo*	idev = Pa_GetDeviceInfo(Pa_GetDefaultInputDevice());
+		const PaDeviceInfo*	odev = Pa_GetDeviceInfo(Pa_GetDefaultOutputDevice());
+
+		fDevNumInChans = (fDsp->getNumInputs() > 0) ? idev->maxInputChannels : 0 ;
+		fDevNumOutChans = (fDsp->getNumOutputs() > 0) ? odev->maxOutputChannels : 0;
+
+		PaStreamParameters inputParameters;
+		PaStreamParameters outputParameters;
+
+		inputParameters.device = Pa_GetDefaultInputDevice();
+		inputParameters.sampleFormat = paFloat32;
+		inputParameters.channelCount = fDevNumInChans;
+		inputParameters.hostApiSpecificStreamInfo = 0;
+
+		outputParameters.device = Pa_GetDefaultOutputDevice();
+		outputParameters.sampleFormat = paFloat32;
+		outputParameters.channelCount = fDevNumOutChans;
+		outputParameters.hostApiSpecificStreamInfo = 0;
+
+		PaError err;
+		if ((err = Pa_IsFormatSupported(
+			((fDevNumInChans > 0) ? &inputParameters : 0),
+			((fDevNumOutChans > 0) ? &outputParameters : 0), fSampleRate)) != 0) {
+			printf("stream format is not supported err = %d\n", err);
+			return false;
+		}
+
+		allocChannels(fBufferSize, max(fDevNumInChans, fDsp->getNumInputs()), max(fDevNumOutChans, fDsp->getNumOutputs()));
+		fDsp->init(fSampleRate);
+		return true;
+	}
+
+	virtual bool start() {
+		pa_error(Pa_OpenDefaultStream(&fAudioStream, fDevNumInChans, fDevNumOutChans, paFloat32, fSampleRate, fBufferSize, audioCallback, this));
+		Pa_StartStream(fAudioStream);
+		return true;
+	}
+
+	virtual void stop() {
+		if (fAudioStream) {
+			Pa_StopStream (fAudioStream);
+			Pa_CloseStream(fAudioStream);
+			fAudioStream = 0;
+		}
+	}
+
+	int processAudio(const float *ibuf, float *obuf, unsigned long frames) {
+		const float* fInputBuffer = ibuf;
+		float* fOutputBuffer = obuf;
+
+		// split input samples
+		for (unsigned long s = 0; s < frames; s++) {
+			for (int c = 0; c < fDevNumInChans; c++) {
+				fInChannel[c][s] = fInputBuffer[c + s*fDevNumInChans];
+			}
+		}
+
+		// process samples
+		fDsp->compute(frames, fInChannel, fOutChannel);
+
+		// merge output samples
+		for (unsigned long s = 0; s < frames; s++) {
+			for (int c = 0; c < fDevNumOutChans; c++) {
+				fOutputBuffer[c + s*fDevNumOutChans] = fOutChannel[c][s];
+			}
+		}
+		return 0;
+	}
+};
+
+//----------------------------------------------------------------------------
+// Port Audio Callback
+//----------------------------------------------------------------------------
+static int audioCallback(const void *ibuf, void *obuf, unsigned long frames, const PaStreamCallbackTimeInfo*,  PaStreamCallbackFlags, void * drv)
+{
+	portaudio * pa = (portaudio*) drv;
+	return pa->processAudio ((const float*)ibuf, (float*)obuf, frames);
+}
+
+#endif