+++ /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