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							- /******************************************/
 - /*
 -   testall.cpp
 -   by Gary P. Scavone, 2007-2008
 - 
 -   This program will make a variety of calls
 -   to extensively test RtAudio functionality.
 - */
 - /******************************************/
 - 
 - #include "RtAudio.h"
 - #include <iostream>
 - #include <cstdlib>
 - #include <cstring>
 - 
 - #define BASE_RATE 0.005
 - #define TIME   1.0
 - 
 - void usage( void ) {
 -   // Error function in case of incorrect command-line
 -   // argument specifications
 -   std::cout << "\nuseage: testall N fs <iDevice> <oDevice> <iChannelOffset> <oChannelOffset>\n";
 -   std::cout << "    where N = number of channels,\n";
 -   std::cout << "    fs = the sample rate,\n";
 -   std::cout << "    iDevice = optional input device to use (default = 0),\n";
 -   std::cout << "    oDevice = optional output device to use (default = 0),\n";
 -   std::cout << "    iChannelOffset = an optional input channel offset (default = 0),\n";
 -   std::cout << "    and oChannelOffset = optional output channel offset (default = 0).\n\n";
 -   exit( 0 );
 - }
 - 
 - unsigned int channels;
 - 
 - // Interleaved buffers
 - int sawi( void *outputBuffer, void * /*inputBuffer*/, unsigned int nBufferFrames,
 -           double /*streamTime*/, RtAudioStreamStatus status, void *data )
 - {
 -   unsigned int i, j;
 -   extern unsigned int channels;
 -   double *buffer = (double *) outputBuffer;
 -   double *lastValues = (double *) data;
 - 
 -   if ( status )
 -     std::cout << "Stream underflow detected!" << std::endl;
 - 
 -   for ( i=0; i<nBufferFrames; i++ ) {
 -     for ( j=0; j<channels; j++ ) {
 -       *buffer++ = (double) lastValues[j];
 -       lastValues[j] += BASE_RATE * (j+1+(j*0.1));
 -       if ( lastValues[j] >= 1.0 ) lastValues[j] -= 2.0;
 -     }
 -   }
 - 
 -   return 0;
 - }
 - 
 - // Non-interleaved buffers
 - int sawni( void *outputBuffer, void * /*inputBuffer*/, unsigned int nBufferFrames,
 -            double /*streamTime*/, RtAudioStreamStatus status, void *data )
 - {
 -   unsigned int i, j;
 -   extern unsigned int channels;
 -   double *buffer = (double *) outputBuffer;
 -   double *lastValues = (double *) data;
 - 
 -   if ( status )
 -     std::cout << "Stream underflow detected!" << std::endl;
 - 
 -   float increment;
 -   for ( j=0; j<channels; j++ ) {
 -     increment = BASE_RATE * (j+1+(j*0.1));
 -     for ( i=0; i<nBufferFrames; i++ ) {
 -       *buffer++ = (double) lastValues[j];
 -       lastValues[j] += increment;
 -       if ( lastValues[j] >= 1.0 ) lastValues[j] -= 2.0;
 -     }
 -   }
 - 
 -   return 0;
 - }
 - 
 - int inout( void *outputBuffer, void *inputBuffer, unsigned int /*nBufferFrames*/,
 -            double /*streamTime*/, RtAudioStreamStatus status, void *data )
 - {
 -   // Since the number of input and output channels is equal, we can do
 -   // a simple buffer copy operation here.
 -   if ( status ) std::cout << "Stream over/underflow detected." << std::endl;
 - 
 -   unsigned int *bytes = (unsigned int *) data;
 -   memcpy( outputBuffer, inputBuffer, *bytes );
 -   return 0;
 - }
 - 
 - int main( int argc, char *argv[] )
 - {
 -   unsigned int bufferFrames, fs, oDevice = 0, iDevice = 0, iOffset = 0, oOffset = 0;
 -   char input;
 - 
 -   // minimal command-line checking
 -   if (argc < 3 || argc > 7 ) usage();
 - 
 -   RtAudio dac;
 -   if ( dac.getDeviceCount() < 1 ) {
 -     std::cout << "\nNo audio devices found!\n";
 -     exit( 1 );
 -   }
 - 
 -   channels = (unsigned int) atoi( argv[1] );
 -   fs = (unsigned int) atoi( argv[2] );
 -   if ( argc > 3 )
 -     iDevice = (unsigned int) atoi( argv[3] );
 -   if ( argc > 4 )
 -     oDevice = (unsigned int) atoi(argv[4]);
 -   if ( argc > 5 )
 -     iOffset = (unsigned int) atoi(argv[5]);
 -   if ( argc > 6 )
 -     oOffset = (unsigned int) atoi(argv[6]);
 - 
 -   double *data = (double *) calloc( channels, sizeof( double ) );
 - 
 -   // Let RtAudio print messages to stderr.
 -   dac.showWarnings( true );
 - 
 -   // Set our stream parameters for output only.
 -   bufferFrames = 512;
 -   RtAudio::StreamParameters oParams, iParams;
 -   oParams.deviceId = oDevice;
 -   oParams.nChannels = channels;
 -   oParams.firstChannel = oOffset;
 - 
 -   if ( oDevice == 0 )
 -     oParams.deviceId = dac.getDefaultOutputDevice();
 - 
 -   RtAudio::StreamOptions options;
 -   options.flags = RTAUDIO_HOG_DEVICE;
 -   try {
 -     dac.openStream( &oParams, NULL, RTAUDIO_FLOAT64, fs, &bufferFrames, &sawi, (void *)data, &options );
 -     std::cout << "\nStream latency = " << dac.getStreamLatency() << std::endl;
 - 
 -     // Start the stream
 -     dac.startStream();
 -     std::cout << "\nPlaying ... press <enter> to stop.\n";
 -     std::cin.get( input );
 - 
 -     // Stop the stream
 -     dac.stopStream();
 - 
 -     // Restart again
 -     std::cout << "Press <enter> to restart.\n";
 -     std::cin.get( input );
 -     dac.startStream();
 - 
 -     // Test abort function
 -     std::cout << "Playing again ... press <enter> to abort.\n";
 -     std::cin.get( input );
 -     dac.abortStream();
 - 
 -     // Restart another time
 -     std::cout << "Press <enter> to restart again.\n";
 -     std::cin.get( input );
 -     dac.startStream();
 - 
 -     std::cout << "Playing again ... press <enter> to close the stream.\n";
 -     std::cin.get( input );
 -   }
 -   catch ( RtAudioError& e ) {
 -     e.printMessage();
 -     goto cleanup;
 -   }
 - 
 -   if ( dac.isStreamOpen() ) dac.closeStream();
 - 
 -   // Test non-interleaved functionality
 -   options.flags = RTAUDIO_NONINTERLEAVED;
 -   try {
 -     dac.openStream( &oParams, NULL, RTAUDIO_FLOAT64, fs, &bufferFrames, &sawni, (void *)data, &options );
 - 
 -     std::cout << "Press <enter> to start non-interleaved playback.\n";
 -     std::cin.get( input );
 - 
 -     // Start the stream
 -     dac.startStream();
 -     std::cout << "\nPlaying ... press <enter> to stop.\n";
 -     std::cin.get( input );
 -   }
 -   catch ( RtAudioError& e ) {
 -     e.printMessage();
 -     goto cleanup;
 -   }
 - 
 -   if ( dac.isStreamOpen() ) dac.closeStream();
 - 
 -   // Now open a duplex stream.
 -   unsigned int bufferBytes;
 -   iParams.deviceId = iDevice;
 -   iParams.nChannels = channels;
 -   iParams.firstChannel = iOffset;
 -   if ( iDevice == 0 )
 -     iParams.deviceId = dac.getDefaultInputDevice();
 -   options.flags = RTAUDIO_NONINTERLEAVED;
 -   try {
 -     dac.openStream( &oParams, &iParams, RTAUDIO_SINT32, fs, &bufferFrames, &inout, (void *)&bufferBytes, &options );
 - 
 -     bufferBytes = bufferFrames * channels * 4;
 - 
 -     std::cout << "Press <enter> to start duplex operation.\n";
 -     std::cin.get( input );
 - 
 -     // Start the stream
 -     dac.startStream();
 -     std::cout << "\nRunning ... press <enter> to stop.\n";
 -     std::cin.get( input );
 - 
 -     // Stop the stream
 -     dac.stopStream();
 -     std::cout << "\nStopped ... press <enter> to restart.\n";
 -     std::cin.get( input );
 - 
 -     // Restart the stream
 -     dac.startStream();
 -     std::cout << "\nRunning ... press <enter> to stop.\n";
 -     std::cin.get( input );
 -   }
 -   catch ( RtAudioError& e ) {
 -     e.printMessage();
 -   }
 - 
 -  cleanup:
 -   if ( dac.isStreamOpen() ) dac.closeStream();
 -   free( data );
 - 
 -   return 0;
 - }
 
 
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