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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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