X-Git-Url: https://main.carlh.net/gitweb/?a=blobdiff_plain;f=tests%2Fplaysaw.cpp;h=d1953b6ca164c0a14ffc3ef136b08d60f5072c26;hb=77cfec0f03ce712410620d93fd698561901bdaf7;hp=b89d5f901bc4a09a0c307a8d0008047b7d594b6d;hpb=1022a7876a6ef1980ad5518340df177814783c7f;p=rtaudio.git

diff --git a/tests/playsaw.cpp b/tests/playsaw.cpp
index b89d5f9..d1953b6 100644
--- a/tests/playsaw.cpp
+++ b/tests/playsaw.cpp
@@ -13,49 +13,74 @@
 #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 )
@@ -74,19 +99,21 @@ int saw( void *outputBuffer, void *inputBuffer, unsigned int nBufferFrames,
 
   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;
@@ -100,12 +127,14 @@ int saw( void *outputBuffer, void *inputBuffer, unsigned int nBufferFrames,
   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
@@ -115,7 +144,7 @@ int main( int argc, char *argv[] )
   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 ) {
@@ -129,6 +158,9 @@ int main( int argc, char *argv[] )
     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 ) );
 
@@ -136,37 +168,45 @@ int main( int argc, char *argv[] )
   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: