--- /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 __alsa_dsp__
+#define __alsa_dsp__
+
+#include <stdio.h>
+#include <pthread.h>
+#include <sys/types.h>
+#include <pwd.h>
+
+#include <alsa/asoundlib.h>
+#include "audio.h"
+#include "dsp.h"
+
+/**
+DEFAULT ALSA PARAMETERS CONTROLLED BY ENVIRONMENT VARIABLES
+
+Some default parameters of Faust's ALSA applications are controlled by the following environment variables :
+
+ FAUST2ALSA_DEVICE = "hw:0"
+ FAUST2ALSA_FREQUENCY= 44100
+ FAUST2ALSA_BUFFER = 1024
+ FAUST2ALSA_PERIODS = 2
+
+*/
+
+using namespace std;
+
+// handle 32/64 bits int size issues
+
+#ifdef __x86_64__
+
+#define uint32 unsigned int
+#define uint64 unsigned long int
+
+#define int32 int
+#define int64 long int
+
+#else
+
+#define uint32 unsigned int
+#define uint64 unsigned long long int
+
+#define int32 int
+#define int64 long long int
+#endif
+
+// check 32/64 bits issues are correctly handled
+
+#define check_error(err) if (err) { printf("%s:%d, alsa error %d : %s\n", __FILE__, __LINE__, err, snd_strerror(err)); exit(1); }
+#define check_error_msg(err,msg) if (err) { fprintf(stderr, "%s:%d, %s : %s(%d)\n", __FILE__, __LINE__, msg, snd_strerror(err), err); exit(1); }
+#define display_error_msg(err,msg) if (err) { fprintf(stderr, "%s:%d, %s : %s(%d)\n", __FILE__, __LINE__, msg, snd_strerror(err), err); }
+
+#define max(x,y) (((x)>(y)) ? (x) : (y))
+#define min(x,y) (((x)<(y)) ? (x) : (y))
+
+/**
+ * Used to set the priority and scheduling of the audi#include <sys/types.h>
+ #include <pwd.h>
+o thread
+ */
+static bool setRealtimePriority ()
+{
+ struct passwd * pw;
+ int err;
+ uid_t uid;
+ struct sched_param param;
+
+ uid = getuid ();
+ pw = getpwnam ("root");
+ setuid (pw->pw_uid);
+ param.sched_priority = 50; /* 0 to 99 */
+ err = sched_setscheduler(0, SCHED_RR, ¶m);
+ setuid (uid);
+ return (err != -1);
+}
+
+/******************************************************************************
+*******************************************************************************
+
+ AUDIO INTERFACE
+
+*******************************************************************************
+*******************************************************************************/
+
+enum { kRead = 1, kWrite = 2, kReadWrite = 3 };
+
+/**
+ * A convenient class to pass parameters to AudioInterface
+ */
+class AudioParam
+{
+ public:
+
+ const char* fCardName;
+ unsigned int fFrequency;
+ unsigned int fBuffering;
+ unsigned int fPeriods;
+
+ unsigned int fSoftInputs;
+ unsigned int fSoftOutputs;
+
+ public :
+ AudioParam() :
+ fCardName("hw:0"),
+ fFrequency(44100),
+ fBuffering(512),
+ fPeriods(2),
+ fSoftInputs(2),
+ fSoftOutputs(2)
+ {}
+
+ AudioParam& cardName(const char* n) { fCardName = n; return *this; }
+ AudioParam& frequency(int f) { fFrequency = f; return *this; }
+ AudioParam& buffering(int fpb) { fBuffering = fpb; return *this; }
+ AudioParam& periods(int p) { fPeriods = p; return *this; }
+ AudioParam& inputs(int n) { fSoftInputs = n; return *this; }
+ AudioParam& outputs(int n) { fSoftOutputs = n; return *this; }
+};
+
+/**
+ * An ALSA audio interface
+ */
+class AudioInterface : public AudioParam
+{
+ public :
+ snd_pcm_t* fOutputDevice ;
+ snd_pcm_t* fInputDevice ;
+ snd_pcm_hw_params_t* fInputParams;
+ snd_pcm_hw_params_t* fOutputParams;
+
+ snd_pcm_format_t fSampleFormat;
+ snd_pcm_access_t fSampleAccess;
+
+ unsigned int fCardInputs;
+ unsigned int fCardOutputs;
+
+ unsigned int fChanInputs;
+ unsigned int fChanOutputs;
+
+ // interleaved mode audiocard buffers
+ void* fInputCardBuffer;
+ void* fOutputCardBuffer;
+
+ // non interleaved mode audiocard buffers
+ void* fInputCardChannels[256];
+ void* fOutputCardChannels[256];
+
+ // non interleaved mod, floating point software buffers
+ float* fInputSoftChannels[256];
+ float* fOutputSoftChannels[256];
+
+ public :
+
+ const char* cardName() { return fCardName; }
+ int frequency() { return fFrequency; }
+ int buffering() { return fBuffering; }
+ int periods() { return fPeriods; }
+
+ float** inputSoftChannels() { return fInputSoftChannels; }
+ float** outputSoftChannels() { return fOutputSoftChannels; }
+
+
+ AudioInterface(const AudioParam& ap = AudioParam()) : AudioParam(ap)
+ {
+
+ fInputDevice = 0;
+ fOutputDevice = 0;
+ fInputParams = 0;
+ fOutputParams = 0;
+ }
+
+ /**
+ * Open the audio interface
+ */
+ void open()
+ {
+ int err;
+
+ // allocation d'un stream d'entree et d'un stream de sortie
+ err = snd_pcm_open( &fInputDevice, fCardName, SND_PCM_STREAM_CAPTURE, 0 ); check_error(err)
+ err = snd_pcm_open( &fOutputDevice, fCardName, SND_PCM_STREAM_PLAYBACK, 0 ); check_error(err)
+
+ // recherche des parametres d'entree
+ err = snd_pcm_hw_params_malloc ( &fInputParams ); check_error(err);
+ setAudioParams(fInputDevice, fInputParams);
+
+ // recherche des parametres de sortie
+ err = snd_pcm_hw_params_malloc ( &fOutputParams ); check_error(err)
+ setAudioParams(fOutputDevice, fOutputParams);
+
+ // set the number of physical input and output channels close to what we need
+ fCardInputs = fSoftInputs;
+ fCardOutputs = fSoftOutputs;
+
+ snd_pcm_hw_params_set_channels_near(fInputDevice, fInputParams, &fCardInputs);
+ snd_pcm_hw_params_set_channels_near(fOutputDevice, fOutputParams, &fCardOutputs);
+
+ printf("inputs : %u, outputs : %u\n", fCardInputs, fCardOutputs);
+
+ // enregistrement des parametres d'entree-sortie
+
+ err = snd_pcm_hw_params (fInputDevice, fInputParams ); check_error (err);
+ err = snd_pcm_hw_params (fOutputDevice, fOutputParams ); check_error (err);
+
+ //assert(snd_pcm_hw_params_get_period_size(fInputParams,NULL) == snd_pcm_hw_params_get_period_size(fOutputParams,NULL));
+
+ // allocation of alsa buffers
+ if (fSampleAccess == SND_PCM_ACCESS_RW_INTERLEAVED) {
+ fInputCardBuffer = calloc(interleavedBufferSize(fInputParams), 1);
+ fOutputCardBuffer = calloc(interleavedBufferSize(fOutputParams), 1);
+
+ } else {
+ for (unsigned int i = 0; i < fCardInputs; i++) {
+ fInputCardChannels[i] = calloc(noninterleavedBufferSize(fInputParams), 1);
+ }
+ for (unsigned int i = 0; i < fCardOutputs; i++) {
+ fOutputCardChannels[i] = calloc(noninterleavedBufferSize(fOutputParams), 1);
+ }
+
+ }
+
+ // allocation of floating point buffers needed by the dsp code
+
+ fChanInputs = max(fSoftInputs, fCardInputs); assert (fChanInputs < 256);
+ fChanOutputs = max(fSoftOutputs, fCardOutputs); assert (fChanOutputs < 256);
+
+ for (unsigned int i = 0; i < fChanInputs; i++) {
+ fInputSoftChannels[i] = (float*) calloc (fBuffering, sizeof(float));
+ for (unsigned int j = 0; j < fBuffering; j++) {
+ fInputSoftChannels[i][j] = 0.0;
+ }
+ }
+
+ for (unsigned int i = 0; i < fChanOutputs; i++) {
+ fOutputSoftChannels[i] = (float*) calloc (fBuffering, sizeof(float));
+ for (unsigned int j = 0; j < fBuffering; j++) {
+ fOutputSoftChannels[i][j] = 0.0;
+ }
+ }
+ }
+
+ void setAudioParams(snd_pcm_t* stream, snd_pcm_hw_params_t* params)
+ {
+ int err;
+
+ // set params record with initial values
+ err = snd_pcm_hw_params_any ( stream, params );
+ check_error_msg(err, "unable to init parameters")
+
+ // set alsa access mode (and fSampleAccess field) either to non interleaved or interleaved
+
+ err = snd_pcm_hw_params_set_access (stream, params, SND_PCM_ACCESS_RW_NONINTERLEAVED );
+ if (err) {
+ err = snd_pcm_hw_params_set_access (stream, params, SND_PCM_ACCESS_RW_INTERLEAVED );
+ check_error_msg(err, "unable to set access mode neither to non-interleaved or to interleaved");
+ }
+ snd_pcm_hw_params_get_access(params, &fSampleAccess);
+
+
+ // search for 32-bits or 16-bits format
+ err = snd_pcm_hw_params_set_format (stream, params, SND_PCM_FORMAT_S32);
+ if (err) {
+ err = snd_pcm_hw_params_set_format (stream, params, SND_PCM_FORMAT_S16);
+ check_error_msg(err, "unable to set format to either 32-bits or 16-bits");
+ }
+ snd_pcm_hw_params_get_format(params, &fSampleFormat);
+ // set sample frequency
+ snd_pcm_hw_params_set_rate_near (stream, params, &fFrequency, 0);
+
+ // set period and period size (buffering)
+ err = snd_pcm_hw_params_set_period_size (stream, params, fBuffering, 0);
+ check_error_msg(err, "period size not available");
+
+ err = snd_pcm_hw_params_set_periods (stream, params, fPeriods, 0);
+ check_error_msg(err, "number of periods not available");
+ }
+
+ ssize_t interleavedBufferSize (snd_pcm_hw_params_t* params)
+ {
+ _snd_pcm_format format; snd_pcm_hw_params_get_format(params, &format);
+ snd_pcm_uframes_t psize; snd_pcm_hw_params_get_period_size(params, &psize, NULL);
+ unsigned int channels; snd_pcm_hw_params_get_channels(params, &channels);
+ ssize_t bsize = snd_pcm_format_size (format, psize * channels);
+ return bsize;
+ }
+
+ ssize_t noninterleavedBufferSize (snd_pcm_hw_params_t* params)
+ {
+ _snd_pcm_format format; snd_pcm_hw_params_get_format(params, &format);
+ snd_pcm_uframes_t psize; snd_pcm_hw_params_get_period_size(params, &psize, NULL);
+ ssize_t bsize = snd_pcm_format_size (format, psize);
+ return bsize;
+ }
+
+ void close()
+ {}
+
+ /**
+ * Read audio samples from the audio card. Convert samples to floats and take
+ * care of interleaved buffers
+ */
+ void read()
+ {
+ if (fSampleAccess == SND_PCM_ACCESS_RW_INTERLEAVED) {
+
+ int count = snd_pcm_readi(fInputDevice, fInputCardBuffer, fBuffering);
+ if (count<0) {
+ display_error_msg(count, "reading samples");
+ int err = snd_pcm_prepare(fInputDevice);
+ check_error_msg(err, "preparing input stream");
+ }
+
+ if (fSampleFormat == SND_PCM_FORMAT_S16) {
+
+ short* buffer16b = (short*) fInputCardBuffer;
+ for (unsigned int s = 0; s < fBuffering; s++) {
+ for (unsigned int c = 0; c < fCardInputs; c++) {
+ fInputSoftChannels[c][s] = float(buffer16b[c + s*fCardInputs])*(1.0/float(SHRT_MAX));
+ }
+ }
+
+ } else if (fSampleFormat == SND_PCM_FORMAT_S32) {
+
+ int32* buffer32b = (int32*) fInputCardBuffer;
+ for (unsigned int s = 0; s < fBuffering; s++) {
+ for (unsigned int c = 0; c < fCardInputs; c++) {
+ fInputSoftChannels[c][s] = float(buffer32b[c + s*fCardInputs])*(1.0/float(INT_MAX));
+ }
+ }
+ } else {
+
+ printf("unrecognized input sample format : %u\n", fSampleFormat);
+ exit(1);
+ }
+
+ } else if (fSampleAccess == SND_PCM_ACCESS_RW_NONINTERLEAVED) {
+
+ int count = snd_pcm_readn(fInputDevice, fInputCardChannels, fBuffering);
+ if (count<0) {
+ display_error_msg(count, "reading samples");
+ int err = snd_pcm_prepare(fInputDevice);
+ check_error_msg(err, "preparing input stream");
+ }
+
+ if (fSampleFormat == SND_PCM_FORMAT_S16) {
+
+ for (unsigned int c = 0; c < fCardInputs; c++) {
+ short* chan16b = (short*) fInputCardChannels[c];
+ for (unsigned int s = 0; s < fBuffering; s++) {
+ fInputSoftChannels[c][s] = float(chan16b[s])*(1.0/float(SHRT_MAX));
+ }
+ }
+
+ } else if (fSampleFormat == SND_PCM_FORMAT_S32) {
+
+ for (unsigned int c = 0; c < fCardInputs; c++) {
+ int32* chan32b = (int32*) fInputCardChannels[c];
+ for (unsigned int s = 0; s < fBuffering; s++) {
+ fInputSoftChannels[c][s] = float(chan32b[s])*(1.0/float(INT_MAX));
+ }
+ }
+ } else {
+
+ printf("unrecognized input sample format : %u\n", fSampleFormat);
+ exit(1);
+ }
+
+ } else {
+ check_error_msg(-10000, "unknow access mode");
+ }
+ }
+
+ /**
+ * write the output soft channels to the audio card. Convert sample
+ * format and interleaves buffers when needed
+ */
+ void write()
+ {
+ recovery :
+
+ if (fSampleAccess == SND_PCM_ACCESS_RW_INTERLEAVED) {
+
+ if (fSampleFormat == SND_PCM_FORMAT_S16) {
+
+ short* buffer16b = (short*) fOutputCardBuffer;
+ for (unsigned int f = 0; f < fBuffering; f++) {
+ for (unsigned int c = 0; c < fCardOutputs; c++) {
+ float x = fOutputSoftChannels[c][f];
+ buffer16b[c + f*fCardOutputs] = short( max(min(x,1.0),-1.0) * float(SHRT_MAX) ) ;
+ }
+ }
+
+ } else if (fSampleFormat == SND_PCM_FORMAT_S32) {
+
+ int32* buffer32b = (int32*) fOutputCardBuffer;
+ for (unsigned int f = 0; f < fBuffering; f++) {
+ for (unsigned int c = 0; c < fCardOutputs; c++) {
+ float x = fOutputSoftChannels[c][f];
+ buffer32b[c + f*fCardOutputs] = int( max(min(x,1.0),-1.0) * float(INT_MAX) ) ;
+ }
+ }
+ } else {
+
+ printf("unrecognized output sample format : %u\n", fSampleFormat);
+ exit(1);
+ }
+
+ int count = snd_pcm_writei(fOutputDevice, fOutputCardBuffer, fBuffering);
+ if (count<0) {
+ display_error_msg(count, "w3");
+ int err = snd_pcm_prepare(fOutputDevice);
+ check_error_msg(err, "preparing output stream");
+ goto recovery;
+ }
+
+
+ } else if (fSampleAccess == SND_PCM_ACCESS_RW_NONINTERLEAVED) {
+
+ if (fSampleFormat == SND_PCM_FORMAT_S16) {
+
+ for (unsigned int c = 0; c < fCardOutputs; c++) {
+ short* chan16b = (short*) fOutputCardChannels[c];
+ for (unsigned int f = 0; f < fBuffering; f++) {
+ float x = fOutputSoftChannels[c][f];
+ chan16b[f] = short( max(min(x,1.0),-1.0) * float(SHRT_MAX) ) ;
+ }
+ }
+
+ } else if (fSampleFormat == SND_PCM_FORMAT_S32) {
+
+ for (unsigned int c = 0; c < fCardOutputs; c++) {
+ int32* chan32b = (int32*) fOutputCardChannels[c];
+ for (unsigned int f = 0; f < fBuffering; f++) {
+ float x = fOutputSoftChannels[c][f];
+ chan32b[f] = int( max(min(x,1.0),-1.0) * float(INT_MAX) ) ;
+ }
+ }
+
+ } else {
+
+ printf("unrecognized output sample format : %u\n", fSampleFormat);
+ exit(1);
+ }
+
+ int count = snd_pcm_writen(fOutputDevice, fOutputCardChannels, fBuffering);
+ if (count<0) {
+ display_error_msg(count, "w3");
+ int err = snd_pcm_prepare(fOutputDevice);
+ check_error_msg(err, "preparing output stream");
+ goto recovery;
+ }
+
+ } else {
+ check_error_msg(-10000, "unknow access mode");
+ }
+ }
+
+ /**
+ * print short information on the audio device
+ */
+ void shortinfo()
+ {
+ int err;
+ snd_ctl_card_info_t* card_info;
+ snd_ctl_t* ctl_handle;
+ err = snd_ctl_open (&ctl_handle, fCardName, 0); check_error(err);
+ snd_ctl_card_info_alloca (&card_info);
+ err = snd_ctl_card_info(ctl_handle, card_info); check_error(err);
+ printf("%s|%d|%d|%d|%d|%s\n",
+ snd_ctl_card_info_get_driver(card_info),
+ fCardInputs, fCardOutputs,
+ fFrequency, fBuffering,
+ snd_pcm_format_name((_snd_pcm_format)fSampleFormat));
+ }
+
+ /**
+ * print more detailled information on the audio device
+ */
+ void longinfo()
+ {
+ int err;
+ snd_ctl_card_info_t* card_info;
+ snd_ctl_t* ctl_handle;
+
+ printf("Audio Interface Description :\n");
+ printf("Sampling Frequency : %d, Sample Format : %s, buffering : %d\n",
+ fFrequency, snd_pcm_format_name((_snd_pcm_format)fSampleFormat), fBuffering);
+ printf("Software inputs : %2d, Software outputs : %2d\n", fSoftInputs, fSoftOutputs);
+ printf("Hardware inputs : %2d, Hardware outputs : %2d\n", fCardInputs, fCardOutputs);
+ printf("Channel inputs : %2d, Channel outputs : %2d\n", fChanInputs, fChanOutputs);
+
+ // affichage des infos de la carte
+ err = snd_ctl_open (&ctl_handle, fCardName, 0); check_error(err);
+ snd_ctl_card_info_alloca (&card_info);
+ err = snd_ctl_card_info(ctl_handle, card_info); check_error(err);
+ printCardInfo(card_info);
+
+ // affichage des infos liees aux streams d'entree-sortie
+ if (fSoftInputs > 0) printHWParams(fInputParams);
+ if (fSoftOutputs > 0) printHWParams(fOutputParams);
+ }
+
+ void printCardInfo(snd_ctl_card_info_t* ci)
+ {
+ printf("Card info (address : %p)\n", ci);
+ printf("\tID = %s\n", snd_ctl_card_info_get_id(ci));
+ printf("\tDriver = %s\n", snd_ctl_card_info_get_driver(ci));
+ printf("\tName = %s\n", snd_ctl_card_info_get_name(ci));
+ printf("\tLongName = %s\n", snd_ctl_card_info_get_longname(ci));
+ printf("\tMixerName = %s\n", snd_ctl_card_info_get_mixername(ci));
+ printf("\tComponents = %s\n", snd_ctl_card_info_get_components(ci));
+ printf("--------------\n");
+ }
+
+ void printHWParams( snd_pcm_hw_params_t* params )
+ {
+ printf("HW Params info (address : %p)\n", params);
+#if 0
+ printf("\tChannels = %d\n", snd_pcm_hw_params_get_channels(params));
+ printf("\tFormat = %s\n", snd_pcm_format_name((_snd_pcm_format)snd_pcm_hw_params_get_format(params)));
+ printf("\tAccess = %s\n", snd_pcm_access_name((_snd_pcm_access)snd_pcm_hw_params_get_access(params)));
+ printf("\tRate = %d\n", snd_pcm_hw_params_get_rate(params, NULL));
+ printf("\tPeriods = %d\n", snd_pcm_hw_params_get_periods(params, NULL));
+ printf("\tPeriod size = %d\n", (int)snd_pcm_hw_params_get_period_size(params, NULL));
+ printf("\tPeriod time = %d\n", snd_pcm_hw_params_get_period_time(params, NULL));
+ printf("\tBuffer size = %d\n", (int)snd_pcm_hw_params_get_buffer_size(params));
+ printf("\tBuffer time = %d\n", snd_pcm_hw_params_get_buffer_time(params, NULL));
+#endif
+ printf("--------------\n");
+ }
+
+};
+
+// lopt : Scan Command Line long int Arguments
+long lopt(int argc, char *argv[], const char* longname, const char* shortname, long def)
+{
+ for (int i=2; i<argc; i++)
+ if ( strcmp(argv[i-1], shortname) == 0 || strcmp(argv[i-1], longname) == 0 )
+ return atoi(argv[i]);
+ return def;
+}
+
+// sopt : Scan Command Line string Arguments
+const char* sopt(int argc, char *argv[], const char* longname, const char* shortname, const char* def)
+{
+ for (int i=2; i<argc; i++)
+ if ( strcmp(argv[i-1], shortname) == 0 || strcmp(argv[i-1], longname) == 0 )
+ return argv[i];
+ return def;
+}
+
+// fopt : Scan Command Line flag option (without argument), return true if the flag
+bool fopt(int argc, char *argv[], const char* longname, const char* shortname)
+{
+ for (int i=1; i<argc; i++)
+ if ( strcmp(argv[i], shortname) == 0 || strcmp(argv[i], longname) == 0 )
+ return true;
+ return false;
+}
+
+/**
+ * Return the value of an environment variable or defval if undefined.
+ */
+static int getDefaultEnv(const char* name, int defval)
+{
+ const char* str = getenv(name);
+ if (str) {
+ return atoi(str);
+ } else {
+ return defval;
+ }
+}
+
+/**
+ * Return the value of an environment variable or defval if undefined.
+ */
+static const char* getDefaultEnv(const char* name, const char* defval)
+{
+ const char* str = getenv(name);
+ if (str) {
+ return str;
+ } else {
+ return defval;
+ }
+}
+
+/******************************************************************************
+*******************************************************************************
+
+ ALSA audio interface
+
+*******************************************************************************
+*******************************************************************************/
+void* __run(void* ptr);
+
+class alsaaudio : public audio {
+ AudioInterface* fAudio;
+ dsp* fDSP;
+ pthread_t fAudioThread;
+ bool fRunning;
+
+ public:
+ alsaaudio(int argc, char *argv[], dsp* DSP) : fAudio(0), fDSP(DSP), fRunning(false) {
+ fAudio = new AudioInterface (
+ AudioParam().cardName( sopt(argc, argv, "--device", "-d", getDefaultEnv("FAUST2ALSA_DEVICE", "hw:0") ) )
+ .frequency( lopt(argc, argv, "--frequency", "-f", getDefaultEnv("FAUST2ALSA_FREQUENCY",44100) ) )
+ .buffering( lopt(argc, argv, "--buffer", "-b", getDefaultEnv("FAUST2ALSA_BUFFER",1024) ) )
+ .periods( lopt(argc, argv, "--periods", "-p", getDefaultEnv("FAUST2ALSA_PERIODS",2) ) )
+ .inputs(DSP->getNumInputs())
+ .outputs(DSP->getNumOutputs()));
+ }
+ virtual ~alsaaudio() { stop(); delete fAudio; }
+
+ virtual bool init(const char */*name*/, dsp* DSP) {
+ AVOIDDENORMALS;
+ fAudio->open();
+ DSP->init(fAudio->frequency());
+ return true;
+ }
+
+ virtual bool start() {
+ fRunning = true;
+ if (pthread_create( &fAudioThread, 0, __run, this))
+ fRunning = false;
+ return fRunning;
+ }
+
+ virtual void stop() {
+ if (fRunning) {
+ fRunning = false;
+ pthread_join (fAudioThread, 0);
+ }
+ }
+
+ virtual void run() {
+ bool rt = setRealtimePriority();
+ printf(rt ? "RT : ":"NRT: "); fAudio->shortinfo();
+ fAudio->write();
+ fAudio->write();
+ while(fRunning) {
+ fAudio->read();
+ fDSP->compute(fAudio->buffering(), fAudio->inputSoftChannels(), fAudio->outputSoftChannels());
+ fAudio->write();
+ }
+ }
+};
+
+void* __run (void* ptr)
+{
+ alsaaudio * alsa = (alsaaudio*)ptr;
+ alsa->run();
+ return 0;
+}
+
+#endif
+
+/********************END ARCHITECTURE SECTION (part 2/2)****************/
+
projects / Faustine.git / blobdiff