--- /dev/null
+/************************************************************************
+
+ 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
projects / Faustine.git / blobdiff