return FAILURE;
}
- // Set the sample rate.
- result = ASIOSetSampleRate( (ASIOSampleRate) sampleRate );
+ // Get the current sample rate
+ ASIOSampleRate currentRate;
+ result = ASIOGetSampleRate( ¤tRate );
if ( result != ASE_OK ) {
drivers.removeCurrentDriver();
- errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error setting sample rate (" << sampleRate << ").";
+ errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error getting sample rate.";
errorText_ = errorStream_.str();
return FAILURE;
}
+ // Set the sample rate only if necessary
+ if ( currentRate != sampleRate ) {
+ result = ASIOSetSampleRate( (ASIOSampleRate) sampleRate );
+ if ( result != ASE_OK ) {
+ drivers.removeCurrentDriver();
+ errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error setting sample rate (" << sampleRate << ").";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
+ }
+
// Determine the driver data type.
ASIOChannelInfo channelInfo;
channelInfo.channel = 0;
stream_.doConvertBuffer[mode] = true;
// Allocate necessary internal buffers
- unsigned long bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat );
+ unsigned long bufferBytes;
+ bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat );
stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 );
if ( stream_.userBuffer[mode] == NULL ) {
errorText_ = "RtApiAsio::probeDeviceOpen: error allocating user buffer memory.";
handle->internalDrain = true;
}
- unsigned int bufferBytes, i, j;
- unsigned int nChannels = stream_.nDeviceChannels[0] + stream_.nDeviceChannels[1];
+ unsigned int nChannels, bufferBytes, i, j;
+ nChannels = stream_.nDeviceChannels[0] + stream_.nDeviceChannels[1];
if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
bufferBytes = stream_.bufferSize * formatBytes( stream_.deviceFormat[0] );
#include <dsound.h>
#include <assert.h>
+#if defined(__MINGW32__)
+// missing from latest mingw winapi
+#define WAVE_FORMAT_96M08 0x00010000 /* 96 kHz, Mono, 8-bit */
+#define WAVE_FORMAT_96S08 0x00020000 /* 96 kHz, Stereo, 8-bit */
+#define WAVE_FORMAT_96M16 0x00040000 /* 96 kHz, Mono, 16-bit */
+#define WAVE_FORMAT_96S16 0x00080000 /* 96 kHz, Stereo, 16-bit */
+#endif
+
#define MINIMUM_DEVICE_BUFFER_SIZE 32768
#ifdef _MSC_VER // if Microsoft Visual C++
// Declarations for utility functions, callbacks, and structures
// specific to the DirectSound implementation.
-static bool CALLBACK deviceCountCallback( LPGUID lpguid,
+static BOOL CALLBACK deviceCountCallback( LPGUID lpguid,
LPCTSTR description,
LPCTSTR module,
LPVOID lpContext );
return s;
}
-static bool CALLBACK deviceCountCallback( LPGUID lpguid,
+static BOOL CALLBACK deviceCountCallback( LPGUID lpguid,
LPCTSTR description,
LPCTSTR module,
LPVOID lpContext )
LPDIRECTSOUNDCAPTURE object;
hr = DirectSoundCaptureCreate( lpguid, &object, NULL );
- if ( hr != DS_OK ) return true;
+ if ( hr != DS_OK ) return TRUE;
caps.dwSize = sizeof(caps);
hr = object->GetCaps( &caps );
DSCAPS caps;
LPDIRECTSOUND object;
hr = DirectSoundCreate( lpguid, &object, NULL );
- if ( hr != DS_OK ) return true;
+ if ( hr != DS_OK ) return TRUE;
caps.dwSize = sizeof(caps);
hr = object->GetCaps( &caps );
object->Release();
}
- if ( info->getDefault && lpguid == NULL ) return false;
+ if ( info->getDefault && lpguid == NULL ) return FALSE;
if ( info->findIndex && info->counter > info->index ) {
info->id = lpguid;
info->name = convertTChar( description );
- return false;
+ return FALSE;
}
- return true;
+ return TRUE;
}
static char* getErrorString( int code )
{
- switch (code) {
+ switch ( code ) {
case DSERR_ALLOCATED:
return "Already allocated";