#include <cstdlib>
/*
-typedef signed long MY_TYPE;
-#define FORMAT RTAUDIO_SINT24
-#define SCALE 2147483647.0
-
-typedef char MY_TYPE;
+typedef char MY_TYPE;
#define FORMAT RTAUDIO_SINT8
#define SCALE 127.0
*/
-typedef signed short MY_TYPE;
+typedef signed short MY_TYPE;
#define FORMAT RTAUDIO_SINT16
#define SCALE 32767.0
/*
-typedef signed long MY_TYPE;
+typedef S24 MY_TYPE;
+#define FORMAT RTAUDIO_SINT24
+#define SCALE 8388607.0
+
+typedef signed long MY_TYPE;
#define FORMAT RTAUDIO_SINT32
#define SCALE 2147483647.0
-typedef float MY_TYPE;
+typedef float MY_TYPE;
#define FORMAT RTAUDIO_FLOAT32
#define SCALE 1.0
-typedef double MY_TYPE;
+typedef double MY_TYPE;
#define FORMAT RTAUDIO_FLOAT64
#define SCALE 1.0
*/
+// Platform-dependent sleep routines.
+#if defined( __WINDOWS_ASIO__ ) || defined( __WINDOWS_DS__ ) || defined( __WINDOWS_WASAPI__ )
+ #include <windows.h>
+ #define SLEEP( milliseconds ) Sleep( (DWORD) milliseconds )
+#else // Unix variants
+ #include <unistd.h>
+ #define SLEEP( milliseconds ) usleep( (unsigned long) (milliseconds * 1000.0) )
+#endif
+
#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: playsaw N fs <device> <channelOffset>\n";
+ std::cout << "\nuseage: playsaw N fs <device> <channelOffset> <time>\n";
std::cout << " where N = number of channels,\n";
std::cout << " fs = the sample rate,\n";
std::cout << " device = optional device to use (default = 0),\n";
- std::cout << " and channelOffset = an optional channel offset on the device (default = 0).\n\n";
+ std::cout << " channelOffset = an optional channel offset on the device (default = 0),\n";
+ std::cout << " and time = an optional time duration in seconds (default = no limit).\n\n";
exit( 0 );
}
+void errorCallback( RtAudioError::Type type, const std::string &errorText )
+{
+ // This example error handling function does exactly the same thing
+ // as the embedded RtAudio::error() function.
+ std::cout << "in errorCallback" << std::endl;
+ if ( type == RtAudioError::WARNING )
+ std::cerr << '\n' << errorText << "\n\n";
+ else if ( type != RtAudioError::WARNING )
+ throw( RtAudioError( errorText, type ) );
+}
+
unsigned int channels;
RtAudio::StreamOptions options;
+unsigned int frameCounter = 0;
+bool checkCount = false;
+unsigned int nFrames = 0;
+const unsigned int callbackReturnValue = 1;
//#define USE_INTERLEAVED
#if defined( USE_INTERLEAVED )
for ( i=0; i<nBufferFrames; i++ ) {
for ( j=0; j<channels; j++ ) {
- *buffer++ = (MY_TYPE) (lastValues[j] * SCALE);
+ *buffer++ = (MY_TYPE) (lastValues[j] * SCALE * 0.5);
lastValues[j] += BASE_RATE * (j+1+(j*0.1));
if ( lastValues[j] >= 1.0 ) lastValues[j] -= 2.0;
}
}
+ frameCounter += nBufferFrames;
+ if ( checkCount && ( frameCounter >= nFrames ) ) return callbackReturnValue;
return 0;
}
#else // Use non-interleaved buffers
-int saw( void *outputBuffer, void *inputBuffer, unsigned int nBufferFrames,
- double streamTime, RtAudioStreamStatus status, void *data )
+int saw( void *outputBuffer, void * /*inputBuffer*/, unsigned int nBufferFrames,
+ double /*streamTime*/, RtAudioStreamStatus status, void *data )
{
unsigned int i, j;
extern unsigned int channels;
for ( j=0; j<channels; j++ ) {
increment = BASE_RATE * (j+1+(j*0.1));
for ( i=0; i<nBufferFrames; i++ ) {
- *buffer++ = (MY_TYPE) (lastValues[j] * SCALE);
+ *buffer++ = (MY_TYPE) (lastValues[j] * SCALE * 0.5);
lastValues[j] += increment;
if ( lastValues[j] >= 1.0 ) lastValues[j] -= 2.0;
}
}
+ frameCounter += nBufferFrames;
+ if ( checkCount && ( frameCounter >= nFrames ) ) return callbackReturnValue;
return 0;
}
#endif
unsigned int bufferFrames, fs, device = 0, offset = 0;
// minimal command-line checking
- if (argc < 3 || argc > 5 ) usage();
+ if (argc < 3 || argc > 6 ) usage();
RtAudio dac;
if ( dac.getDeviceCount() < 1 ) {
device = (unsigned int) atoi( argv[3] );
if ( argc > 4 )
offset = (unsigned int) atoi( argv[4] );
+ if ( argc > 5 )
+ nFrames = (unsigned int) (fs * atof( argv[5] ));
+ if ( nFrames > 0 ) checkCount = true;
double *data = (double *) calloc( channels, sizeof( double ) );
dac.showWarnings( true );
// Set our stream parameters for output only.
- bufferFrames = 256;
+ bufferFrames = 512;
RtAudio::StreamParameters oParams;
oParams.deviceId = device;
oParams.nChannels = channels;
oParams.firstChannel = offset;
- options.flags |= RTAUDIO_HOG_DEVICE;
+ if ( device == 0 )
+ oParams.deviceId = dac.getDefaultOutputDevice();
+
+ options.flags = RTAUDIO_HOG_DEVICE;
options.flags |= RTAUDIO_SCHEDULE_REALTIME;
#if !defined( USE_INTERLEAVED )
options.flags |= RTAUDIO_NONINTERLEAVED;
#endif
try {
- dac.openStream( &oParams, NULL, FORMAT, fs, &bufferFrames, &saw, (void *)data, &options );
+ dac.openStream( &oParams, NULL, FORMAT, fs, &bufferFrames, &saw, (void *)data, &options, &errorCallback );
dac.startStream();
}
- catch ( RtError& e ) {
+ catch ( RtAudioError& e ) {
e.printMessage();
goto cleanup;
}
- char input;
- //std::cout << "Stream latency = " << dac.getStreamLatency() << "\n" << std::endl;
- std::cout << "\nPlaying ... press <enter> to quit (buffer size = " << bufferFrames << ").\n";
- std::cin.get( input );
-
- try {
- // Stop the stream
- dac.stopStream();
+ if ( checkCount ) {
+ while ( dac.isStreamRunning() == true ) SLEEP( 100 );
}
- catch ( RtError& e ) {
- e.printMessage();
+ else {
+ char input;
+ //std::cout << "Stream latency = " << dac.getStreamLatency() << "\n" << std::endl;
+ std::cout << "\nPlaying ... press <enter> to quit (buffer size = " << bufferFrames << ").\n";
+ std::cin.get( input );
+
+ try {
+ // Stop the stream
+ dac.stopStream();
+ }
+ catch ( RtAudioError& e ) {
+ e.printMessage();
+ }
}
cleanup:
projects / rtaudio-cdist.git / blobdiff