mirror of
https://github.com/thestk/stk
synced 2026-01-11 20:11:52 +00:00
Updates to eguitar, ragamatic and examples project files for new RtAudio API.
This commit is contained in:
@@ -11,76 +11,107 @@
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#include <iostream>
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#include <map>
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int main()
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{
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// Create an api map.
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std::map<int, std::string> apiMap;
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apiMap[RtAudio::MACOSX_CORE] = "OS-X CoreAudio";
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apiMap[RtAudio::WINDOWS_ASIO] = "Windows ASIO";
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apiMap[RtAudio::WINDOWS_DS] = "Windows DirectSound";
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apiMap[RtAudio::UNIX_JACK] = "Jack Client";
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apiMap[RtAudio::LINUX_ALSA] = "Linux ALSA";
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apiMap[RtAudio::LINUX_OSS] = "Linux OSS";
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apiMap[RtAudio::RTAUDIO_DUMMY] = "RtAudio Dummy";
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void usage( void ) {
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// Error function in case of incorrect command-line
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// argument specifications
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std::cout << "\nuseage: audioprobe <apiname> <nRepeats>\n";
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std::cout << " where apiname = an optional api (ex., 'core', default = all compiled),\n";
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std::cout << " and nRepeats = an optional number of times to repeat the device query (default = 0),\n";
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std::cout << " which can be used to test device (dis)connections.\n\n";
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exit( 0 );
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}
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std::vector< RtAudio::Api > listApis()
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{
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std::vector< RtAudio::Api > apis;
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RtAudio :: getCompiledApi( apis );
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std::cout << "\nCompiled APIs:\n";
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for ( unsigned int i=0; i<apis.size(); i++ )
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std::cout << " " << apiMap[ apis[i] ] << std::endl;
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for ( size_t i=0; i<apis.size(); i++ )
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std::cout << i << ". " << RtAudio::getApiDisplayName(apis[i])
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<< " (" << RtAudio::getApiName(apis[i]) << ")" << std::endl;
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RtAudio audio;
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return apis;
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}
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void listDevices( RtAudio& audio )
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{
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RtAudio::DeviceInfo info;
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std::cout << "\nCurrent API: " << apiMap[ audio.getCurrentApi() ] << std::endl;
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std::cout << "\nAPI: " << RtAudio::getApiDisplayName(audio.getCurrentApi()) << std::endl;
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unsigned int devices = audio.getDeviceCount();
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std::cout << "\nFound " << devices << " device(s) ...\n";
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std::vector<unsigned int> devices = audio.getDeviceIds();
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std::cout << "\nFound " << devices.size() << " device(s) ...\n";
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for (unsigned int i=0; i<devices; i++) {
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info = audio.getDeviceInfo(i);
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for (unsigned int i=0; i<devices.size(); i++) {
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info = audio.getDeviceInfo( devices[i] );
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std::cout << "\nDevice Name = " << info.name << '\n';
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if ( info.probed == false )
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std::cout << "Probe Status = UNsuccessful\n";
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std::cout << "Device Index = " << i << '\n';
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std::cout << "Output Channels = " << info.outputChannels << '\n';
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std::cout << "Input Channels = " << info.inputChannels << '\n';
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std::cout << "Duplex Channels = " << info.duplexChannels << '\n';
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if ( info.isDefaultOutput ) std::cout << "This is the default output device.\n";
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else std::cout << "This is NOT the default output device.\n";
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if ( info.isDefaultInput ) std::cout << "This is the default input device.\n";
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else std::cout << "This is NOT the default input device.\n";
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if ( info.nativeFormats == 0 )
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std::cout << "No natively supported data formats(?)!";
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else {
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std::cout << "Probe Status = Successful\n";
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std::cout << "Output Channels = " << info.outputChannels << '\n';
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std::cout << "Input Channels = " << info.inputChannels << '\n';
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std::cout << "Duplex Channels = " << info.duplexChannels << '\n';
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if ( info.isDefaultOutput ) std::cout << "This is the default output device.\n";
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else std::cout << "This is NOT the default output device.\n";
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if ( info.isDefaultInput ) std::cout << "This is the default input device.\n";
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else std::cout << "This is NOT the default input device.\n";
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if ( info.nativeFormats == 0 )
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std::cout << "No natively supported data formats(?)!";
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else {
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std::cout << "Natively supported data formats:\n";
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if ( info.nativeFormats & RTAUDIO_SINT8 )
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std::cout << " 8-bit int\n";
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if ( info.nativeFormats & RTAUDIO_SINT16 )
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std::cout << " 16-bit int\n";
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if ( info.nativeFormats & RTAUDIO_SINT24 )
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std::cout << " 24-bit int\n";
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if ( info.nativeFormats & RTAUDIO_SINT32 )
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std::cout << " 32-bit int\n";
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if ( info.nativeFormats & RTAUDIO_FLOAT32 )
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std::cout << " 32-bit float\n";
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if ( info.nativeFormats & RTAUDIO_FLOAT64 )
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std::cout << " 64-bit float\n";
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std::cout << "Natively supported data formats:\n";
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if ( info.nativeFormats & RTAUDIO_SINT8 )
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std::cout << " 8-bit int\n";
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if ( info.nativeFormats & RTAUDIO_SINT16 )
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std::cout << " 16-bit int\n";
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if ( info.nativeFormats & RTAUDIO_SINT24 )
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std::cout << " 24-bit int\n";
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if ( info.nativeFormats & RTAUDIO_SINT32 )
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std::cout << " 32-bit int\n";
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if ( info.nativeFormats & RTAUDIO_FLOAT32 )
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std::cout << " 32-bit float\n";
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if ( info.nativeFormats & RTAUDIO_FLOAT64 )
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std::cout << " 64-bit float\n";
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}
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if ( info.sampleRates.size() < 1 )
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std::cout << "No supported sample rates found!";
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else {
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std::cout << "Supported sample rates = ";
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for (unsigned int j=0; j<info.sampleRates.size(); j++)
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std::cout << info.sampleRates[j] << " ";
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}
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std::cout << std::endl;
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if ( info.preferredSampleRate == 0 )
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std::cout << "No preferred sample rate found!" << std::endl;
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else
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std::cout << "Preferred sample rate = " << info.preferredSampleRate << std::endl;
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}
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}
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int main(int argc, char *argv[])
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{
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std::cout << "\nRtAudio Version " << RtAudio::getVersion() << std::endl;
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std::vector< RtAudio::Api > apis = listApis();
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// minimal command-line checking
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if (argc > 3 ) usage();
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unsigned int nRepeats = 0;
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if ( argc > 2 ) nRepeats = (unsigned int) atoi( argv[2] );
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char input;
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for ( size_t api=0; api < apis.size(); api++ ) {
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if (argc < 2 || apis[api] == RtAudio::getCompiledApiByName(argv[1]) ) {
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RtAudio audio(apis[api]);
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for ( size_t n=0; n <= nRepeats; n++ ) {
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listDevices(audio);
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if ( n < nRepeats ) {
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std::cout << std::endl;
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std::cout << "\nWaiting ... press <enter> to repeat.\n";
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std::cin.get(input);
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}
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}
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if ( info.sampleRates.size() < 1 )
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std::cout << "No supported sample rates found!";
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else {
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std::cout << "Supported sample rates = ";
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for (unsigned int j=0; j<info.sampleRates.size(); j++)
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std::cout << info.sampleRates[j] << " ";
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}
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std::cout << std::endl;
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}
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}
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std::cout << std::endl;
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return 0;
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}
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@@ -56,11 +56,8 @@ int main()
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parameters.nChannels = 1;
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RtAudioFormat format = ( sizeof(StkFloat) == 8 ) ? RTAUDIO_FLOAT64 : RTAUDIO_FLOAT32;
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unsigned int bufferFrames = RT_BUFFER_SIZE;
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try {
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dac.openStream( ¶meters, NULL, format, (unsigned int)Stk::sampleRate(), &bufferFrames, &tick, (void *)&data );
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}
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catch ( RtAudioError& error ) {
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error.printMessage();
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if ( dac.openStream( ¶meters, NULL, format, (unsigned int)Stk::sampleRate(), &bufferFrames, &tick, (void *)&data ) ) {
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std::cout << dac.getErrorText() << std::endl;
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goto cleanup;
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}
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@@ -75,11 +72,8 @@ int main()
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data.frequency = 220.0;
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data.instrument->noteOn( data.frequency, 0.5 );
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try {
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dac.startStream();
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}
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catch ( RtAudioError &error ) {
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error.printMessage();
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if ( dac.startStream() ) {
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std::cout << dac.getErrorText() << std::endl;
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goto cleanup;
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}
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@@ -88,12 +82,7 @@ int main()
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Stk::sleep( 100 );
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// Shut down the callback and output stream.
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try {
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dac.closeStream();
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}
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catch ( RtAudioError &error ) {
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error.printMessage();
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}
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dac.closeStream();
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cleanup:
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delete data.instrument;
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@@ -130,11 +130,8 @@ int main( int argc, char *argv[] )
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parameters.nChannels = 1;
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RtAudioFormat format = ( sizeof(StkFloat) == 8 ) ? RTAUDIO_FLOAT64 : RTAUDIO_FLOAT32;
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unsigned int bufferFrames = RT_BUFFER_SIZE;
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try {
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dac.openStream( ¶meters, NULL, format, (unsigned int)Stk::sampleRate(), &bufferFrames, &tick, (void *)&data );
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}
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catch ( RtAudioError &error ) {
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error.printMessage();
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if ( dac.openStream( ¶meters, NULL, format, (unsigned int)Stk::sampleRate(), &bufferFrames, &tick, (void *)&data ) ) {
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std::cout << dac.getErrorText() << std::endl;
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goto cleanup;
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}
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@@ -149,11 +146,8 @@ int main( int argc, char *argv[] )
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if ( data.messager.setScoreFile( argv[1] ) == false )
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goto cleanup;
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try {
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dac.startStream();
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}
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catch ( RtAudioError &error ) {
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error.printMessage();
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if ( dac.startStream() ) {
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std::cout << dac.getErrorText() << std::endl;
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goto cleanup;
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}
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@@ -162,12 +156,7 @@ int main( int argc, char *argv[] )
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Stk::sleep( 100 );
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// Shut down the output stream.
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try {
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dac.closeStream();
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}
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catch ( RtAudioError &error ) {
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error.printMessage();
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}
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dac.closeStream();
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cleanup:
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delete data.instrument;
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@@ -33,21 +33,15 @@ int main()
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parameters.nChannels = 1;
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RtAudioFormat format = ( sizeof(StkFloat) == 8 ) ? RTAUDIO_FLOAT64 : RTAUDIO_FLOAT32;
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unsigned int bufferFrames = RT_BUFFER_SIZE;
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try {
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dac.openStream( ¶meters, NULL, format, (unsigned int)Stk::sampleRate(), &bufferFrames, &tick, (void *)&sine );
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}
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catch ( RtAudioError &error ) {
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error.printMessage();
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if ( dac.openStream( ¶meters, NULL, format, (unsigned int)Stk::sampleRate(), &bufferFrames, &tick, (void *)&sine ) ) {
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std::cout << dac.getErrorText() << std::endl;
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goto cleanup;
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}
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sine.setFrequency(440.0);
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try {
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dac.startStream();
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}
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catch ( RtAudioError &error ) {
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error.printMessage();
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if ( dac.startStream() ) {
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std::cout << dac.getErrorText() << std::endl;
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goto cleanup;
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}
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@@ -57,12 +51,7 @@ int main()
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std::cin.get( keyhit );
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// Shut down the output stream.
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try {
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dac.closeStream();
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}
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catch ( RtAudioError &error ) {
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error.printMessage();
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}
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dac.closeStream();
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cleanup:
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@@ -1,7 +1,7 @@
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/******************************************/
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/*
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duplex.cpp
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by Gary P. Scavone, 2006-2007.
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by Gary P. Scavone, 2006-2019.
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This program opens a duplex stream and passes
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input directly through to the output.
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@@ -14,24 +14,26 @@
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#include <cstring>
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/*
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typedef signed long MY_TYPE;
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#define FORMAT RTAUDIO_SINT24
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typedef char MY_TYPE;
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typedef char MY_TYPE;
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#define FORMAT RTAUDIO_SINT8
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typedef signed short MY_TYPE;
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#define FORMAT RTAUDIO_SINT16
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typedef signed long MY_TYPE;
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#define FORMAT RTAUDIO_SINT32
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typedef float MY_TYPE;
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#define FORMAT RTAUDIO_FLOAT32
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*/
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typedef double MY_TYPE;
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typedef signed short MY_TYPE;
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#define FORMAT RTAUDIO_SINT16
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/*
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typedef S24 MY_TYPE;
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#define FORMAT RTAUDIO_SINT24
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typedef signed long MY_TYPE;
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#define FORMAT RTAUDIO_SINT32
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typedef float MY_TYPE;
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#define FORMAT RTAUDIO_FLOAT32
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typedef double MY_TYPE;
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#define FORMAT RTAUDIO_FLOAT64
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*/
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void usage( void ) {
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// Error function in case of incorrect command-line
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@@ -39,26 +41,52 @@ void usage( void ) {
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std::cout << "\nuseage: duplex N fs <iDevice> <oDevice> <iChannelOffset> <oChannelOffset>\n";
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std::cout << " where N = number of channels,\n";
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std::cout << " fs = the sample rate,\n";
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std::cout << " iDevice = optional input device to use (default = 0),\n";
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std::cout << " oDevice = optional output device to use (default = 0),\n";
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std::cout << " iDevice = optional input device index to use (default = 0),\n";
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std::cout << " oDevice = optional output device index to use (default = 0),\n";
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std::cout << " iChannelOffset = an optional input channel offset (default = 0),\n";
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std::cout << " and oChannelOffset = optional output channel offset (default = 0).\n\n";
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exit( 0 );
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}
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int inout( void *outputBuffer, void *inputBuffer, unsigned int nBufferFrames,
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unsigned int getDeviceIndex( std::vector<std::string> deviceNames, bool isInput = false )
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{
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unsigned int i;
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std::string keyHit;
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std::cout << '\n';
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for ( i=0; i<deviceNames.size(); i++ )
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std::cout << " Device #" << i << ": " << deviceNames[i] << '\n';
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do {
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if ( isInput )
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std::cout << "\nChoose an input device #: ";
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else
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std::cout << "\nChoose an output device #: ";
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std::cin >> i;
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} while ( i >= deviceNames.size() );
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std::getline( std::cin, keyHit ); // used to clear out stdin
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return i;
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}
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double streamTimePrintIncrement = 1.0; // seconds
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double streamTimePrintTime = 1.0; // seconds
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int inout( void *outputBuffer, void *inputBuffer, unsigned int /*nBufferFrames*/,
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double streamTime, RtAudioStreamStatus status, void *data )
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{
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// Since the number of input and output channels is equal, we can do
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// a simple buffer copy operation here.
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if ( status ) std::cout << "Stream over/underflow detected." << std::endl;
|
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unsigned long *bytes = (unsigned long *) data;
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if ( streamTime >= streamTimePrintTime ) {
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std::cout << "streamTime = " << streamTime << std::endl;
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streamTimePrintTime += streamTimePrintIncrement;
|
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}
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unsigned int *bytes = (unsigned int *) data;
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memcpy( outputBuffer, inputBuffer, *bytes );
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return 0;
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}
|
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|
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int main(int argc, char *argv[])
|
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int main( int argc, char *argv[] )
|
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{
|
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unsigned int channels, fs, bufferBytes, oDevice = 0, iDevice = 0, iOffset = 0, oOffset = 0;
|
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@@ -66,9 +94,10 @@ int main(int argc, char *argv[])
|
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if (argc < 3 || argc > 7 ) usage();
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RtAudio adac;
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if ( adac.getDeviceCount() < 1 ) {
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std::vector<unsigned int> deviceIds = adac.getDeviceIds();
|
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if ( deviceIds.size() < 1 ) {
|
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std::cout << "\nNo audio devices found!\n";
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exit( 0 );
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exit( 1 );
|
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}
|
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|
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channels = (unsigned int) atoi(argv[1]);
|
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@@ -88,48 +117,48 @@ int main(int argc, char *argv[])
|
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// Set the same number of channels for both input and output.
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unsigned int bufferFrames = 512;
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RtAudio::StreamParameters iParams, oParams;
|
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if ( iDevice == 0 )
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iParams.deviceId = adac.getDefaultInputDevice();
|
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else
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iParams.deviceId = iDevice - 1;
|
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iParams.nChannels = channels;
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iParams.firstChannel = iOffset;
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if ( oDevice == 0 )
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oParams.deviceId = adac.getDefaultOutputDevice();
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else
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oParams.deviceId = oDevice - 1;
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oParams.nChannels = channels;
|
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oParams.firstChannel = oOffset;
|
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|
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if ( iDevice == 0 )
|
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iParams.deviceId = adac.getDefaultInputDevice();
|
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else {
|
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if ( iDevice >= deviceIds.size() )
|
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iDevice = getDeviceIndex( adac.getDeviceNames(), true );
|
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iParams.deviceId = deviceIds[iDevice];
|
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}
|
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if ( oDevice == 0 )
|
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oParams.deviceId = adac.getDefaultOutputDevice();
|
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else {
|
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if ( oDevice >= deviceIds.size() )
|
||||
oDevice = getDeviceIndex( adac.getDeviceNames() );
|
||||
oParams.deviceId = deviceIds[oDevice];
|
||||
}
|
||||
|
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RtAudio::StreamOptions options;
|
||||
//options.flags |= RTAUDIO_NONINTERLEAVED;
|
||||
|
||||
bufferBytes = bufferFrames * channels * sizeof( MY_TYPE );
|
||||
try {
|
||||
adac.openStream( &oParams, &iParams, FORMAT, fs, &bufferFrames, &inout, (void *)&bufferBytes, &options );
|
||||
}
|
||||
catch ( RtAudioError& e ) {
|
||||
std::cout << '\n' << e.getMessage() << '\n' << std::endl;
|
||||
exit( 1 );
|
||||
if ( adac.openStream( &oParams, &iParams, FORMAT, fs, &bufferFrames, &inout, (void *)&bufferBytes, &options ) ) {
|
||||
goto cleanup;
|
||||
}
|
||||
|
||||
if ( adac.isStreamOpen() == false ) goto cleanup;
|
||||
|
||||
// Test RtAudio functionality for reporting latency.
|
||||
std::cout << "\nStream latency = " << adac.getStreamLatency() << " frames" << std::endl;
|
||||
|
||||
try {
|
||||
adac.startStream();
|
||||
if ( adac.startStream() ) goto cleanup;
|
||||
|
||||
char input;
|
||||
std::cout << "\nRunning ... press <enter> to quit (buffer frames = " << bufferFrames << ").\n";
|
||||
std::cin.get(input);
|
||||
char input;
|
||||
std::cout << "\nRunning ... press <enter> to quit (buffer frames = " << bufferFrames << ").\n";
|
||||
std::cin.get(input);
|
||||
|
||||
// Stop the stream.
|
||||
// Stop the stream.
|
||||
if ( adac.isStreamRunning() )
|
||||
adac.stopStream();
|
||||
}
|
||||
catch ( RtAudioError& e ) {
|
||||
std::cout << '\n' << e.getMessage() << '\n' << std::endl;
|
||||
goto cleanup;
|
||||
}
|
||||
|
||||
cleanup:
|
||||
if ( adac.isStreamOpen() ) adac.closeStream();
|
||||
|
||||
@@ -79,35 +79,23 @@ int main( int argc, char *argv[] )
|
||||
parameters.nChannels = grani.channelsOut();
|
||||
RtAudioFormat format = ( sizeof(StkFloat) == 8 ) ? RTAUDIO_FLOAT64 : RTAUDIO_FLOAT32;
|
||||
unsigned int bufferFrames = RT_BUFFER_SIZE;
|
||||
try {
|
||||
dac.openStream( ¶meters, NULL, format, (unsigned int)Stk::sampleRate(), &bufferFrames, &tick, (void *)&grani );
|
||||
}
|
||||
catch ( RtAudioError &error ) {
|
||||
error.printMessage();
|
||||
if ( dac.openStream( ¶meters, NULL, format, (unsigned int)Stk::sampleRate(), &bufferFrames, &tick, (void *)&grani ) ) {
|
||||
std::cout << dac.getErrorText() << std::endl;
|
||||
goto cleanup;
|
||||
}
|
||||
|
||||
try {
|
||||
dac.startStream();
|
||||
}
|
||||
catch ( RtAudioError &error ) {
|
||||
error.printMessage();
|
||||
if ( dac.startStream() ) {
|
||||
std::cout << dac.getErrorText() << std::endl;
|
||||
goto cleanup;
|
||||
}
|
||||
|
||||
|
||||
// Block waiting here.
|
||||
char keyhit;
|
||||
std::cout << "\nPlaying ... press <enter> to quit.\n";
|
||||
std::cin.get( keyhit );
|
||||
|
||||
// Shut down the callback and output stream.
|
||||
try {
|
||||
dac.closeStream();
|
||||
}
|
||||
catch ( RtAudioError &error ) {
|
||||
error.printMessage();
|
||||
}
|
||||
dac.closeStream();
|
||||
|
||||
cleanup:
|
||||
|
||||
|
||||
@@ -99,11 +99,8 @@ int main(int argc, char *argv[])
|
||||
parameters.nChannels = ( channels == 1 ) ? 2 : channels; // Play mono files as stereo.
|
||||
RtAudioFormat format = ( sizeof(StkFloat) == 8 ) ? RTAUDIO_FLOAT64 : RTAUDIO_FLOAT32;
|
||||
unsigned int bufferFrames = RT_BUFFER_SIZE;
|
||||
try {
|
||||
dac.openStream( ¶meters, NULL, format, (unsigned int)Stk::sampleRate(), &bufferFrames, &tick, (void *)&input );
|
||||
}
|
||||
catch ( RtAudioError &error ) {
|
||||
error.printMessage();
|
||||
if ( dac.openStream( ¶meters, NULL, format, (unsigned int)Stk::sampleRate(), &bufferFrames, &tick, (void *)&input ) ) {
|
||||
std::cout << dac.getErrorText() << std::endl;
|
||||
goto cleanup;
|
||||
}
|
||||
|
||||
@@ -113,11 +110,8 @@ int main(int argc, char *argv[])
|
||||
// Resize the StkFrames object appropriately.
|
||||
frames.resize( bufferFrames, channels );
|
||||
|
||||
try {
|
||||
dac.startStream();
|
||||
}
|
||||
catch ( RtAudioError &error ) {
|
||||
error.printMessage();
|
||||
if ( dac.startStream() ) {
|
||||
std::cout << dac.getErrorText() << std::endl;
|
||||
goto cleanup;
|
||||
}
|
||||
|
||||
@@ -127,12 +121,7 @@ int main(int argc, char *argv[])
|
||||
|
||||
// By returning a non-zero value in the callback above, the stream
|
||||
// is automatically stopped. But we should still close it.
|
||||
try {
|
||||
dac.closeStream();
|
||||
}
|
||||
catch ( RtAudioError &error ) {
|
||||
error.printMessage();
|
||||
}
|
||||
dac.closeStream();
|
||||
|
||||
cleanup:
|
||||
return 0;
|
||||
|
||||
@@ -130,11 +130,8 @@ int main()
|
||||
parameters.nChannels = 1;
|
||||
RtAudioFormat format = ( sizeof(StkFloat) == 8 ) ? RTAUDIO_FLOAT64 : RTAUDIO_FLOAT32;
|
||||
unsigned int bufferFrames = RT_BUFFER_SIZE;
|
||||
try {
|
||||
dac.openStream( ¶meters, NULL, format, (unsigned int)Stk::sampleRate(), &bufferFrames, &tick, (void *)&data );
|
||||
}
|
||||
catch ( RtAudioError &error ) {
|
||||
error.printMessage();
|
||||
if ( dac.openStream( ¶meters, NULL, format, (unsigned int)Stk::sampleRate(), &bufferFrames, &tick, (void *)&data ) ) {
|
||||
std::cout << dac.getErrorText() << std::endl;
|
||||
goto cleanup;
|
||||
}
|
||||
|
||||
@@ -154,11 +151,8 @@ int main()
|
||||
if ( data.messager.startStdInput() == false )
|
||||
goto cleanup;
|
||||
|
||||
try {
|
||||
dac.startStream();
|
||||
}
|
||||
catch ( RtAudioError &error ) {
|
||||
error.printMessage();
|
||||
if ( dac.startStream() ) {
|
||||
std::cout << dac.getErrorText() << std::endl;
|
||||
goto cleanup;
|
||||
}
|
||||
|
||||
@@ -167,12 +161,7 @@ int main()
|
||||
Stk::sleep( 100 );
|
||||
|
||||
// Shut down the callback and output stream.
|
||||
try {
|
||||
dac.closeStream();
|
||||
}
|
||||
catch ( RtAudioError &error ) {
|
||||
error.printMessage();
|
||||
}
|
||||
dac.closeStream();
|
||||
|
||||
cleanup:
|
||||
for ( i=0; i<3; i++ ) delete instrument[i];
|
||||
|
||||
Reference in New Issue
Block a user