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* * |
* * |
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* LinuxSampler - modular, streaming capable sampler * |
* LinuxSampler - modular, streaming capable sampler * |
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* * |
* * |
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* Copyright (C) 2003 by Benno Senoner and Christian Schoenebeck * |
* Copyright (C) 2003, 2004 by Benno Senoner and Christian Schoenebeck * |
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* * |
* * |
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* This program is free software; you can redistribute it and/or modify * |
* This program is free software; you can redistribute it and/or modify * |
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* it under the terms of the GNU General Public License as published by * |
* it under the terms of the GNU General Public License as published by * |
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* MA 02111-1307 USA * |
* MA 02111-1307 USA * |
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***************************************************************************/ |
***************************************************************************/ |
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#include <stdio.h> |
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#include <stdlib.h> |
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#include <unistd.h> |
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#include <getopt.h> |
#include <getopt.h> |
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#include <signal.h> |
#include <signal.h> |
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#include <pthread.h> |
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#include <string> |
#include "Sampler.h" |
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#include "audiodriver/AudioOutputDeviceFactory.h" |
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#include "global.h" |
#include "network/lscpserver.h" |
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#include "audioio.h" |
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#include "diskthread.h" |
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#include "audiothread.h" |
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#include "midiin.h" |
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#include "stream.h" |
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#include "RIFF.h" |
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#include "gig.h" |
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#if 0 |
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#define AUDIO_CHANNELS 2 // stereo |
#define AUDIO_CHANNELS 2 // stereo |
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#define AUDIO_FRAGMENTS 3 // 3 fragments, if it does not work set it to 2 |
#define AUDIO_FRAGMENTS 3 // 3 fragments, if it does not work set it to 2 |
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#define AUDIO_FRAGMENTSIZE 512 // each fragment has 512 frames |
#define AUDIO_FRAGMENTSIZE 512 // each fragment has 512 frames |
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#define AUDIO_SAMPLERATE 44100 // Hz |
#define AUDIO_SAMPLERATE 44100 // Hz |
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#endif |
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using namespace LinuxSampler; |
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enum patch_format_t { |
/*enum patch_format_t { |
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patch_format_unknown, |
patch_format_unknown, |
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patch_format_gig, |
patch_format_gig, |
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patch_format_dls |
patch_format_dls |
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} patch_format; |
} patch_format = patch_format_unknown;*/ |
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Sampler* pSampler = NULL; |
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LSCPServer* pLSCPServer = NULL; |
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pthread_t signalhandlerthread; |
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/*AudioThread* pEngine = NULL; |
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uint instrument_index = 0; |
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double volume = 0.25; |
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int num_fragments = AUDIO_FRAGMENTS; |
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int fragmentsize = AUDIO_FRAGMENTSIZE; |
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uint samplerate = AUDIO_SAMPLERATE; |
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String input_client; |
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String alsaout = "0,0"; // default card |
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String jack_playback[2] = { "", "" }; |
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bool use_jack = true; |
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bool run_server = false;*/ |
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AudioIO* pAudioIO; |
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DiskThread* pDiskThread; |
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AudioThread* pAudioThread; |
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MidiIn* pMidiInThread; |
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RIFF::File* pRIFF; |
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gig::File* pGig; |
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gig::Instrument* pInstrument; |
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uint instrument_index; |
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double volume; |
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int num_fragments; |
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int fragmentsize; |
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std::string input_client; |
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pthread_t signalhandlerthread; |
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void parse_options(int argc, char **argv); |
void parse_options(int argc, char **argv); |
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void signal_handler(int signal); |
void signal_handler(int signal); |
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int main(int argc, char **argv) { |
int main(int argc, char **argv) { |
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pAudioIO = NULL; |
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pRIFF = NULL; |
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pGig = NULL; |
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// setting signal handler for catching SIGINT (thus e.g. <CTRL><C>) |
// setting signal handler for catching SIGINT (thus e.g. <CTRL><C>) |
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signalhandlerthread = pthread_self(); |
signalhandlerthread = pthread_self(); |
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signal(SIGINT, signal_handler); |
signal(SIGINT, signal_handler); |
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patch_format = patch_format_unknown; |
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instrument_index = 0; |
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num_fragments = AUDIO_FRAGMENTS; |
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fragmentsize = AUDIO_FRAGMENTSIZE; |
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volume = 0.25; // default volume |
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// parse and assign command line options |
// parse and assign command line options |
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parse_options(argc, argv); |
//parse_options(argc, argv); |
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if (patch_format != patch_format_gig) { |
/*if (patch_format != patch_format_gig) { |
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printf("Sorry only Gigasampler loading migrated in LinuxSampler so far, use --gig to load a .gig file!\n"); |
printf("Sorry only Gigasampler loading migrated in LinuxSampler so far, use --gig to load a .gig file!\n"); |
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printf("Use 'linuxsampler --help' to see all available options.\n"); |
printf("Use 'linuxsampler --help' to see all available options.\n"); |
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return EXIT_FAILURE; |
return EXIT_FAILURE; |
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} |
}*/ |
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dmsg(1,("Initializing audio output...")); |
// create LinuxSampler instance |
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pAudioIO = new AudioIO(); |
dmsg(1,("Creating Sampler...")); |
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int error = pAudioIO->Initialize(AUDIO_CHANNELS, AUDIO_SAMPLERATE, num_fragments, fragmentsize); |
pSampler = new Sampler; |
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if (error) return EXIT_FAILURE; |
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dmsg(1,("OK\n")); |
dmsg(1,("OK\n")); |
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// Loading gig file |
dmsg(1,("Registered audio output drivers: %s\n", AudioOutputDeviceFactory::AvailableDriversAsString().c_str())); |
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try { |
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printf("Loading gig file..."); |
// create an audio output device |
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fflush(stdout); |
/* bool no_jack = true; |
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pRIFF = new RIFF::File(argv[argc - 1]); |
#if HAVE_JACK |
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pGig = new gig::File(pRIFF); |
if (use_jack) { |
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pInstrument = pGig->GetInstrument(instrument_index); |
dmsg(1,("Creating audio output device (Jack)...")); |
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if (!pInstrument) { |
try { |
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printf("there's no instrument with index %d.\n", instrument_index); |
pSampler->CreateAudioOutputDevice(audio_output_type_jack); |
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exit(EXIT_FAILURE); |
no_jack = false; |
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} |
} |
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pGig->GetFirstSample(); // just to complete instrument loading before we enter the realtime part |
catch (AudioOutputException aoe) { |
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printf("OK\n"); |
aoe.PrintMessage(); |
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fflush(stdout); |
dmsg(1,("Trying to create Alsa output device instead.\n")); |
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} |
} |
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catch (RIFF::Exception e) { |
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e.PrintMessage(); |
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return EXIT_FAILURE; |
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} |
} |
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catch (...) { |
#endif // HAVE_JACK |
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printf("Unknown exception while trying to parse gig file.\n"); |
if (no_jack) { |
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return EXIT_FAILURE; |
dmsg(1,("Creating audio output device (Alsa)...")); |
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try { |
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pSampler->CreateAudioOutputDevice(audio_output_type_alsa); |
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} |
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catch (AudioOutputException aoe) { |
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aoe.PrintMessage(); |
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dmsg(1,("Trying to create Alsa output device instead.\n")); |
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return EXIT_FAILURE; |
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} |
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} |
} |
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dmsg(1,("OK\n"));*/ |
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DiskThread* pDiskThread = new DiskThread(((pAudioIO->FragmentSize << MAX_PITCH) << 1) + 6); //FIXME: assuming stereo |
// start LSCP network server |
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AudioThread* pAudioThread = new AudioThread(pAudioIO, pDiskThread, pInstrument); |
dmsg(1,("Starting network server...")); |
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MidiIn* pMidiInThread = new MidiIn(pAudioThread); |
pLSCPServer = new LSCPServer(pSampler); |
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pLSCPServer->StartThread(); |
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dmsg(1,("Starting disk thread...")); |
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pDiskThread->StartThread(); |
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dmsg(1,("OK\n")); |
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dmsg(1,("Starting MIDI in thread...")); |
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if (input_client.size() > 0) pMidiInThread->SubscribeToClient(input_client.c_str()); |
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pMidiInThread->StartThread(); |
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dmsg(1,("OK\n")); |
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sleep(1); |
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dmsg(1,("Starting audio thread...")); |
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pAudioThread->Volume = volume; |
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pAudioThread->StartThread(); |
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dmsg(1,("OK\n")); |
dmsg(1,("OK\n")); |
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printf("LinuxSampler initialization completed.\n"); |
printf("LinuxSampler initialization completed.\n"); |
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while(true) { |
while(true) { |
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printf("Voices: %3.3d (Max: %3.3d) Streams: %3.3d (Max: %3.3d, Unused: %3.3d)\r", |
/*printf("Voices: %3.3d (Max: %3.3d) Streams: %3.3d (Max: %3.3d, Unused: %3.3d)\r", |
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pAudioThread->ActiveVoiceCount, pAudioThread->ActiveVoiceCountMax, |
pEngine->ActiveVoiceCount, pEngine->ActiveVoiceCountMax, |
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pDiskThread->ActiveStreamCount, pDiskThread->ActiveStreamCountMax, Stream::GetUnusedStreams()); |
pEngine->pDiskThread->ActiveStreamCount, pEngine->pDiskThread->ActiveStreamCountMax, Stream::GetUnusedStreams()); |
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fflush(stdout); |
fflush(stdout);*/ |
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usleep(500000); |
usleep(500000); |
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} |
} |
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void signal_handler(int signal) { |
void signal_handler(int signal) { |
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if (pthread_equal(pthread_self(), signalhandlerthread) && signal == SIGINT) { |
if (pthread_equal(pthread_self(), signalhandlerthread) && signal == SIGINT) { |
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// stop all threads |
if (pLSCPServer) { |
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if (pMidiInThread) pMidiInThread->StopThread(); |
pLSCPServer->StopThread(); |
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if (pAudioThread) pAudioThread->StopThread(); |
delete pLSCPServer; |
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if (pDiskThread) pDiskThread->StopThread(); |
} |
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if (pSampler) delete pSampler; |
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// free all resources |
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if (pMidiInThread) delete pMidiInThread; |
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if (pAudioThread) delete pAudioThread; |
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if (pDiskThread) delete pDiskThread; |
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if (pGig) delete pGig; |
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if (pRIFF) delete pRIFF; |
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if (pAudioIO) delete pAudioIO; |
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printf("LinuxSampler stopped due to SIGINT\n"); |
printf("LinuxSampler stopped due to SIGINT\n"); |
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exit(EXIT_SUCCESS); |
exit(EXIT_SUCCESS); |
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} |
} |
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} |
} |
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void parse_options(int argc, char **argv) { |
/*void parse_options(int argc, char **argv) { |
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int res; |
int res; |
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int option_index = 0; |
int option_index = 0; |
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static struct option long_options[] = |
static struct option long_options[] = |
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{"gig",0,0,0}, |
{"gig",0,0,0}, |
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{"instrument",1,0,0}, |
{"instrument",1,0,0}, |
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{"inputclient",1,0,0}, |
{"inputclient",1,0,0}, |
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{"alsaout",1,0,0}, |
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{"jackout",1,0,0}, |
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{"samplerate",1,0,0}, |
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{"server",0,0,0}, |
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{"help",0,0,0}, |
{"help",0,0,0}, |
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{0,0,0,0} |
{0,0,0,0} |
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}; |
}; |
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if(res == -1) break; |
if(res == -1) break; |
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if (res == 0) { |
if (res == 0) { |
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switch(option_index) { |
switch(option_index) { |
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case 0: |
case 0: // --numfragments |
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num_fragments = atoi(optarg); |
num_fragments = atoi(optarg); |
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break; |
break; |
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case 1: |
case 1: // --fragmentsize |
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fragmentsize = atoi(optarg); |
fragmentsize = atoi(optarg); |
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break; |
break; |
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case 2: |
case 2: // --volume |
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volume = atof(optarg); |
volume = atof(optarg); |
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break; |
break; |
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case 3: |
case 3: // --dls |
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patch_format = patch_format_dls; |
patch_format = patch_format_dls; |
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break; |
break; |
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case 4: |
case 4: // --gig |
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patch_format = patch_format_gig; |
patch_format = patch_format_gig; |
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break; |
break; |
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case 5: |
case 5: // --instrument |
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instrument_index = atoi(optarg); |
instrument_index = atoi(optarg); |
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break; |
break; |
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case 6: |
case 6: // --inputclient |
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input_client = optarg; |
input_client = optarg; |
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break; |
break; |
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case 7: |
case 7: // --alsaout |
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alsaout = optarg; |
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use_jack = false; // If this option is specified do not connect to jack |
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break; |
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case 8: { // --jackout |
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try { |
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String arg(optarg); |
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// remove outer apostrophes |
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arg = arg.substr(arg.find('\'') + 1, arg.rfind('\'') - (arg.find('\'') + 1)); |
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// split in two arguments |
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jack_playback[0] = arg.substr(0, arg.find("\' ")); |
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jack_playback[1] = arg.substr(arg.find("\' ") + 2, arg.size() - (arg.find("\' ") + 2)); |
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// remove inner apostrophes |
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jack_playback[0] = jack_playback[0].substr(0, jack_playback[0].find('\'')); |
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jack_playback[1] = jack_playback[1].substr(jack_playback[1].find('\'') + 1, jack_playback[1].size() - jack_playback[1].find('\'')); |
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// this is the default but set it up anyway in case alsa_card was also used. |
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use_jack = true; |
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} |
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catch (...) { |
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fprintf(stderr, "Invalid argument '%s' for parameter --jackout\n", optarg); |
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exit(EXIT_FAILURE); |
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} |
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break; |
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} |
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case 9: // --samplerate |
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samplerate = atoi(optarg); |
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break; |
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case 10: // --server |
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run_server = true; |
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break; |
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case 11: // --help |
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printf("usage: linuxsampler [OPTIONS] <INSTRUMENTFILE>\n\n"); |
printf("usage: linuxsampler [OPTIONS] <INSTRUMENTFILE>\n\n"); |
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printf("--gig loads a Gigasampler instrument\n"); |
printf("--gig loads a Gigasampler instrument\n"); |
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printf("--dls loads a DLS instrument\n"); |
printf("--dls loads a DLS instrument\n"); |
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printf(" default: 0.25)\n"); |
printf(" default: 0.25)\n"); |
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printf("--inputclient connects to an Alsa sequencer input client on startup\n"); |
printf("--inputclient connects to an Alsa sequencer input client on startup\n"); |
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printf(" (e.g. 64:0 to connect to a client with ID 64 and port 0)\n"); |
printf(" (e.g. 64:0 to connect to a client with ID 64 and port 0)\n"); |
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exit(0); |
printf("--alsaout connects to the given Alsa sound device on startup\n"); |
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printf(" (e.g. 0,0 to connect to hw:0,0 or plughw:0,0)\n"); |
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printf("--jackout connects to the given Jack playback ports on startup\n"); |
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printf(" (e.g. \"\'alsa_pcm:playback_1\' \'alsa_pcm:playback_2\'\"\n"); |
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printf(" in case of stereo output)\n"); |
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printf("--samplerate sets sample rate if supported by audio output system\n"); |
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printf(" (e.g. 44100)\n"); |
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printf("--server launch network server for remote control\n"); |
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exit(EXIT_SUCCESS); |
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break; |
break; |
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} |
} |
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} |
} |
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} |
} |
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} |
}*/ |