RtAudio WWW site: http://www.music.mcgill.ca/~gary/rtaudio/
RtAudio: realtime audio i/o C++ classes
- Copyright (c) 2001-2008 Gary P. Scavone
+ Copyright (c) 2001-2010 Gary P. Scavone
Permission is hereby granted, free of charge, to any person
obtaining a copy of this software and associated documentation files
*/
/************************************************************************/
-// RtAudio: Version 4.0.4
+// RtAudio: Version 4.0.7
#include "RtAudio.h"
#include <iostream>
+#include <cstdlib>
+#include <cstring>
+#include <climits>
// Static variable definitions.
const unsigned int RtApi::MAX_SAMPLE_RATES = 14;
// implementation.
struct CoreHandle {
AudioDeviceID id[2]; // device ids
+#if defined( MAC_OS_X_VERSION_10_5 ) && ( MAC_OS_X_VERSION_MIN_REQUIRED >= MAC_OS_X_VERSION_10_5 )
AudioDeviceIOProcID procId[2];
+#endif
UInt32 iStream[2]; // device stream index (or first if using multiple)
UInt32 nStreams[2]; // number of streams to use
bool xrun[2];
{
// Find out how many audio devices there are, if any.
UInt32 dataSize;
- OSStatus result = AudioHardwareGetPropertyInfo( kAudioHardwarePropertyDevices, &dataSize, NULL );
+ AudioObjectPropertyAddress propertyAddress = { kAudioHardwarePropertyDevices, kAudioObjectPropertyScopeGlobal, kAudioObjectPropertyElementMaster };
+ OSStatus result = AudioObjectGetPropertyDataSize( kAudioObjectSystemObject, &propertyAddress, 0, NULL, &dataSize );
if ( result != noErr ) {
errorText_ = "RtApiCore::getDeviceCount: OS-X error getting device info!";
error( RtError::WARNING );
AudioDeviceID id;
UInt32 dataSize = sizeof( AudioDeviceID );
- OSStatus result = AudioHardwareGetProperty( kAudioHardwarePropertyDefaultInputDevice,
- &dataSize, &id );
-
+ AudioObjectPropertyAddress property = { kAudioHardwarePropertyDefaultInputDevice, kAudioObjectPropertyScopeGlobal, kAudioObjectPropertyElementMaster };
+ OSStatus result = AudioObjectGetPropertyData( kAudioObjectSystemObject, &property, 0, NULL, &dataSize, &id );
if ( result != noErr ) {
errorText_ = "RtApiCore::getDefaultInputDevice: OS-X system error getting device.";
error( RtError::WARNING );
dataSize *= nDevices;
AudioDeviceID deviceList[ nDevices ];
- result = AudioHardwareGetProperty( kAudioHardwarePropertyDevices, &dataSize, (void *) &deviceList );
+ property.mSelector = kAudioHardwarePropertyDevices;
+ result = AudioObjectGetPropertyData( kAudioObjectSystemObject, &property, 0, NULL, &dataSize, (void *) &deviceList );
if ( result != noErr ) {
errorText_ = "RtApiCore::getDefaultInputDevice: OS-X system error getting device IDs.";
error( RtError::WARNING );
AudioDeviceID id;
UInt32 dataSize = sizeof( AudioDeviceID );
- OSStatus result = AudioHardwareGetProperty( kAudioHardwarePropertyDefaultOutputDevice,
- &dataSize, &id );
-
+ AudioObjectPropertyAddress property = { kAudioHardwarePropertyDefaultOutputDevice, kAudioObjectPropertyScopeGlobal, kAudioObjectPropertyElementMaster };
+ OSStatus result = AudioObjectGetPropertyData( kAudioObjectSystemObject, &property, 0, NULL, &dataSize, &id );
if ( result != noErr ) {
errorText_ = "RtApiCore::getDefaultOutputDevice: OS-X system error getting device.";
error( RtError::WARNING );
return 0;
}
- dataSize *= nDevices;
+ dataSize = sizeof( AudioDeviceID ) * nDevices;
AudioDeviceID deviceList[ nDevices ];
- result = AudioHardwareGetProperty( kAudioHardwarePropertyDevices, &dataSize, (void *) &deviceList );
+ property.mSelector = kAudioHardwarePropertyDevices;
+ result = AudioObjectGetPropertyData( kAudioObjectSystemObject, &property, 0, NULL, &dataSize, (void *) &deviceList );
if ( result != noErr ) {
errorText_ = "RtApiCore::getDefaultOutputDevice: OS-X system error getting device IDs.";
error( RtError::WARNING );
AudioDeviceID deviceList[ nDevices ];
UInt32 dataSize = sizeof( AudioDeviceID ) * nDevices;
- OSStatus result = AudioHardwareGetProperty( kAudioHardwarePropertyDevices, &dataSize, (void *) &deviceList );
+ AudioObjectPropertyAddress property = { kAudioHardwarePropertyDevices,
+ kAudioObjectPropertyScopeGlobal,
+ kAudioObjectPropertyElementMaster };
+ OSStatus result = AudioObjectGetPropertyData( kAudioObjectSystemObject, &property,
+ 0, NULL, &dataSize, (void *) &deviceList );
if ( result != noErr ) {
errorText_ = "RtApiCore::getDeviceInfo: OS-X system error getting device IDs.";
error( RtError::WARNING );
// Get the device name.
info.name.erase();
- char name[256];
- dataSize = 256;
- result = AudioDeviceGetProperty( id, 0, false,
- kAudioDevicePropertyDeviceManufacturer,
- &dataSize, name );
-
+ CFStringRef cfname;
+ dataSize = sizeof( CFStringRef );
+ property.mSelector = kAudioObjectPropertyManufacturer;
+ result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, &cfname );
if ( result != noErr ) {
errorStream_ << "RtApiCore::probeDeviceInfo: system error (" << getErrorCode( result ) << ") getting device manufacturer.";
errorText_ = errorStream_.str();
error( RtError::WARNING );
return info;
}
- info.name.append( (const char *)name, strlen(name) );
+
+ const char *mname = CFStringGetCStringPtr( cfname, CFStringGetSystemEncoding() );
+ info.name.append( (const char *)mname, strlen(mname) );
info.name.append( ": " );
+ CFRelease( cfname );
- dataSize = 256;
- result = AudioDeviceGetProperty( id, 0, false,
- kAudioDevicePropertyDeviceName,
- &dataSize, name );
+ property.mSelector = kAudioObjectPropertyName;
+ result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, &cfname );
if ( result != noErr ) {
errorStream_ << "RtApiCore::probeDeviceInfo: system error (" << getErrorCode( result ) << ") getting device name.";
errorText_ = errorStream_.str();
error( RtError::WARNING );
return info;
}
+
+ const char *name = CFStringGetCStringPtr( cfname, CFStringGetSystemEncoding() );
info.name.append( (const char *)name, strlen(name) );
+ CFRelease( cfname );
// Get the output stream "configuration".
AudioBufferList *bufferList = nil;
- result = AudioDeviceGetPropertyInfo( id, 0, false,
- kAudioDevicePropertyStreamConfiguration,
- &dataSize, NULL );
- if (result != noErr || dataSize == 0) {
+ property.mSelector = kAudioDevicePropertyStreamConfiguration;
+ property.mScope = kAudioDevicePropertyScopeOutput;
+ // property.mElement = kAudioObjectPropertyElementWildcard;
+ dataSize = 0;
+ result = AudioObjectGetPropertyDataSize( id, &property, 0, NULL, &dataSize );
+ if ( result != noErr || dataSize == 0 ) {
errorStream_ << "RtApiCore::getDeviceInfo: system error (" << getErrorCode( result ) << ") getting output stream configuration info for device (" << device << ").";
errorText_ = errorStream_.str();
error( RtError::WARNING );
return info;
}
- result = AudioDeviceGetProperty( id, 0, false,
- kAudioDevicePropertyStreamConfiguration,
- &dataSize, bufferList );
- if ( result != noErr ) {
+ result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, bufferList );
+ if ( result != noErr || dataSize == 0 ) {
free( bufferList );
errorStream_ << "RtApiCore::getDeviceInfo: system error (" << getErrorCode( result ) << ") getting output stream configuration for device (" << device << ").";
errorText_ = errorStream_.str();
free( bufferList );
// Get the input stream "configuration".
- result = AudioDeviceGetPropertyInfo( id, 0, true,
- kAudioDevicePropertyStreamConfiguration,
- &dataSize, NULL );
- if (result != noErr || dataSize == 0) {
+ property.mScope = kAudioDevicePropertyScopeInput;
+ result = AudioObjectGetPropertyDataSize( id, &property, 0, NULL, &dataSize );
+ if ( result != noErr || dataSize == 0 ) {
errorStream_ << "RtApiCore::getDeviceInfo: system error (" << getErrorCode( result ) << ") getting input stream configuration info for device (" << device << ").";
errorText_ = errorStream_.str();
error( RtError::WARNING );
return info;
}
- result = AudioDeviceGetProperty( id, 0, true,
- kAudioDevicePropertyStreamConfiguration,
- &dataSize, bufferList );
- if ( result != noErr ) {
+ result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, bufferList );
+ if (result != noErr || dataSize == 0) {
free( bufferList );
errorStream_ << "RtApiCore::getDeviceInfo: system error (" << getErrorCode( result ) << ") getting input stream configuration for device (" << device << ").";
errorText_ = errorStream_.str();
if ( info.outputChannels == 0 ) isInput = true;
// Determine the supported sample rates.
- result = AudioDeviceGetPropertyInfo( id, 0, isInput,
- kAudioDevicePropertyAvailableNominalSampleRates,
- &dataSize, NULL );
-
+ property.mSelector = kAudioDevicePropertyAvailableNominalSampleRates;
+ if ( isInput == false ) property.mScope = kAudioDevicePropertyScopeOutput;
+ result = AudioObjectGetPropertyDataSize( id, &property, 0, NULL, &dataSize );
if ( result != kAudioHardwareNoError || dataSize == 0 ) {
errorStream_ << "RtApiCore::getDeviceInfo: system error (" << getErrorCode( result ) << ") getting sample rate info.";
errorText_ = errorStream_.str();
UInt32 nRanges = dataSize / sizeof( AudioValueRange );
AudioValueRange rangeList[ nRanges ];
- result = AudioDeviceGetProperty( id, 0, isInput,
- kAudioDevicePropertyAvailableNominalSampleRates,
- &dataSize, &rangeList );
-
+ result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, &rangeList );
if ( result != kAudioHardwareNoError ) {
errorStream_ << "RtApiCore::getDeviceInfo: system error (" << getErrorCode( result ) << ") getting sample rates.";
errorText_ = errorStream_.str();
// no interest to the client.
info.nativeFormats = RTAUDIO_FLOAT32;
- if ( getDefaultOutputDevice() == device )
- info.isDefaultOutput = true;
- if ( getDefaultInputDevice() == device )
- info.isDefaultInput = true;
+ if ( info.outputChannels > 0 )
+ if ( getDefaultOutputDevice() == device ) info.isDefaultOutput = true;
+ if ( info.inputChannels > 0 )
+ if ( getDefaultInputDevice() == device ) info.isDefaultInput = true;
info.probed = true;
return info;
return kAudioHardwareNoError;
}
-OSStatus deviceListener( AudioDeviceID inDevice,
- UInt32 channel,
- Boolean isInput,
- AudioDevicePropertyID propertyID,
+OSStatus deviceListener( AudioObjectID inDevice,
+ UInt32 nAddresses,
+ const AudioObjectPropertyAddress properties[],
void* handlePointer )
{
CoreHandle *handle = (CoreHandle *) handlePointer;
- if ( propertyID == kAudioDeviceProcessorOverload ) {
- if ( isInput )
- handle->xrun[1] = true;
- else
- handle->xrun[0] = true;
+ for ( UInt32 i=0; i<nAddresses; i++ ) {
+ if ( properties[i].mSelector == kAudioDeviceProcessorOverload ) {
+ if ( properties[i].mScope == kAudioDevicePropertyScopeInput )
+ handle->xrun[1] = true;
+ else
+ handle->xrun[0] = true;
+ }
}
return kAudioHardwareNoError;
}
-static bool hasProperty( AudioDeviceID id, UInt32 channel, bool isInput, AudioDevicePropertyID property )
-{
- OSStatus result = AudioDeviceGetPropertyInfo( id, channel, isInput, property, NULL, NULL );
- return result == 0;
-}
-
bool RtApiCore :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,
unsigned int firstChannel, unsigned int sampleRate,
RtAudioFormat format, unsigned int *bufferSize,
AudioDeviceID deviceList[ nDevices ];
UInt32 dataSize = sizeof( AudioDeviceID ) * nDevices;
- OSStatus result = AudioHardwareGetProperty( kAudioHardwarePropertyDevices, &dataSize, (void *) &deviceList );
+ AudioObjectPropertyAddress property = { kAudioHardwarePropertyDevices,
+ kAudioObjectPropertyScopeGlobal,
+ kAudioObjectPropertyElementMaster };
+ OSStatus result = AudioObjectGetPropertyData( kAudioObjectSystemObject, &property,
+ 0, NULL, &dataSize, (void *) &deviceList );
if ( result != noErr ) {
errorText_ = "RtApiCore::probeDeviceOpen: OS-X system error getting device IDs.";
return FAILURE;
// Setup for stream mode.
bool isInput = false;
- if ( mode == INPUT ) isInput = true;
-
- // Set or disable "hog" mode.
- dataSize = sizeof( UInt32 );
- UInt32 doHog = 0;
- if ( options && options->flags & RTAUDIO_HOG_DEVICE ) doHog = 1;
- result = AudioHardwareSetProperty( kAudioHardwarePropertyHogModeIsAllowed, dataSize, &doHog );
- if ( result != noErr ) {
- errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") setting 'hog' state!";
- errorText_ = errorStream_.str();
- return FAILURE;
+ if ( mode == INPUT ) {
+ isInput = true;
+ property.mScope = kAudioDevicePropertyScopeInput;
}
+ else
+ property.mScope = kAudioDevicePropertyScopeOutput;
// Get the stream "configuration".
- AudioBufferList *bufferList;
- result = AudioDeviceGetPropertyInfo( id, 0, isInput,
- kAudioDevicePropertyStreamConfiguration,
- &dataSize, NULL );
- if (result != noErr || dataSize == 0) {
+ AudioBufferList *bufferList = nil;
+ dataSize = 0;
+ property.mSelector = kAudioDevicePropertyStreamConfiguration;
+ result = AudioObjectGetPropertyDataSize( id, &property, 0, NULL, &dataSize );
+ if ( result != noErr || dataSize == 0 ) {
errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting stream configuration info for device (" << device << ").";
errorText_ = errorStream_.str();
return FAILURE;
return FAILURE;
}
- result = AudioDeviceGetProperty( id, 0, isInput,
- kAudioDevicePropertyStreamConfiguration,
- &dataSize, bufferList );
- if ( result != noErr ) {
- free( bufferList );
+ result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, bufferList );
+ if (result != noErr || dataSize == 0) {
errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting stream configuration for device (" << device << ").";
errorText_ = errorStream_.str();
return FAILURE;
free( bufferList );
- std::cout << "deviceStreams = " << nStreams << ", firstStream = " << firstStream << ", streamCount = " << streamCount << ", channelOffset = " << channelOffset << std::endl;
-
// Determine the buffer size.
AudioValueRange bufferRange;
dataSize = sizeof( AudioValueRange );
- result = AudioDeviceGetProperty( id, 0, isInput,
- kAudioDevicePropertyBufferFrameSizeRange,
- &dataSize, &bufferRange );
+ property.mSelector = kAudioDevicePropertyBufferFrameSizeRange;
+ result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, &bufferRange );
+
if ( result != noErr ) {
errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting buffer size range for device (" << device << ").";
errorText_ = errorStream_.str();
// need to make this setting for the master channel.
UInt32 theSize = (UInt32) *bufferSize;
dataSize = sizeof( UInt32 );
- result = AudioDeviceSetProperty( id, NULL, 0, isInput,
- kAudioDevicePropertyBufferFrameSize,
- dataSize, &theSize );
+ property.mSelector = kAudioDevicePropertyBufferFrameSize;
+ result = AudioObjectSetPropertyData( id, &property, 0, NULL, dataSize, &theSize );
if ( result != noErr ) {
errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") setting the buffer size for device (" << device << ").";
stream_.bufferSize = *bufferSize;
stream_.nBuffers = 1;
- // Get the stream ID(s) so we can set the stream format. We'll have
- // to do this for each stream.
+ // Check and if necessary, change the sample rate for the device.
+ Float64 nominalRate;
+ dataSize = sizeof( Float64 );
+ property.mSelector = kAudioDevicePropertyNominalSampleRate;
+ result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, &nominalRate );
+
+ if ( result != noErr ) {
+ errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting current sample rate.";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
+
+ // Only change the sample rate if off by more than 1 Hz.
+ if ( fabs( nominalRate - (double)sampleRate ) > 1.0 ) {
+ nominalRate = (Float64) sampleRate;
+ result = AudioObjectSetPropertyData( id, &property, 0, NULL, dataSize, &nominalRate );
+
+ if ( result != noErr ) {
+ errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") setting sample rate for device (" << device << ").";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
+ }
+
+ // Try to set "hog" mode ... it's not clear to me this is working.
+ if ( options && options->flags & RTAUDIO_HOG_DEVICE ) {
+ pid_t hog_pid;
+ dataSize = sizeof( hog_pid );
+ property.mSelector = kAudioDevicePropertyHogMode;
+ result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, &hog_pid );
+ if ( result != noErr ) {
+ errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting 'hog' state!";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
+
+ if ( hog_pid != getpid() ) {
+ hog_pid = getpid();
+ result = AudioObjectSetPropertyData( id, &property, 0, NULL, dataSize, &hog_pid );
+ if ( result != noErr ) {
+ errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") setting 'hog' state!";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
+ }
+ }
+
+ // Get the stream ID(s) so we can set the stream format.
AudioStreamID streamIDs[ nStreams ];
dataSize = nStreams * sizeof( AudioStreamID );
- result = AudioDeviceGetProperty( id, 0, isInput,
- kAudioDevicePropertyStreams,
- &dataSize, &streamIDs );
+ property.mSelector = kAudioDevicePropertyStreams;
+ result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, &streamIDs );
+
if ( result != noErr ) {
errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting stream ID(s) for device (" << device << ").";
errorText_ = errorStream_.str();
return FAILURE;
}
- // Now set the stream format. Also, check the physical format of the
- // device and change that if necessary.
+ // Now set the stream format for each stream. Also, check the
+ // physical format of the device and change that if necessary.
AudioStreamBasicDescription description;
dataSize = sizeof( AudioStreamBasicDescription );
bool updateFormat;
for ( UInt32 i=0; i<streamCount; i++ ) {
- result = AudioStreamGetProperty( streamIDs[firstStream+i], 0,
- kAudioStreamPropertyVirtualFormat,
- &dataSize, &description );
+ property.mSelector = kAudioStreamPropertyVirtualFormat;
+ result = AudioObjectGetPropertyData( streamIDs[firstStream+i], &property, 0, NULL, &dataSize, &description );
if ( result != noErr ) {
errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting stream format for device (" << device << ").";
}
if ( updateFormat ) {
- result = AudioStreamSetProperty( streamIDs[firstStream+i], NULL, 0,
- kAudioStreamPropertyVirtualFormat,
- dataSize, &description );
+ result = AudioObjectSetPropertyData( streamIDs[firstStream+i], &property, 0, NULL, dataSize, &description );
if ( result != noErr ) {
errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") setting sample rate or data format for device (" << device << ").";
errorText_ = errorStream_.str();
}
// Now check the physical format.
- result = AudioStreamGetProperty( streamIDs[firstStream+i], 0,
- kAudioStreamPropertyPhysicalFormat,
- &dataSize, &description );
+ property.mSelector = kAudioStreamPropertyPhysicalFormat;
+ result = AudioObjectGetPropertyData( streamIDs[firstStream+i], &property, 0, NULL, &dataSize, &description );
if ( result != noErr ) {
errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting stream physical format for device (" << device << ").";
errorText_ = errorStream_.str();
// We'll try higher bit rates first and then work our way down.
testDescription.mBitsPerChannel = 32;
+ testDescription.mBytesPerFrame = testDescription.mBitsPerChannel/8 * testDescription.mChannelsPerFrame;
+ testDescription.mBytesPerPacket = testDescription.mBytesPerFrame * testDescription.mFramesPerPacket;
formatFlags = description.mFormatFlags | kLinearPCMFormatFlagIsFloat & ~kLinearPCMFormatFlagIsSignedInteger;
testDescription.mFormatFlags = formatFlags;
- result = AudioStreamSetProperty( streamIDs[firstStream+i], NULL, 0, kAudioStreamPropertyPhysicalFormat, dataSize, &testDescription );
+ result = AudioObjectSetPropertyData( streamIDs[firstStream+i], &property, 0, NULL, dataSize, &testDescription );
if ( result == noErr ) continue;
testDescription = description;
testDescription.mBitsPerChannel = 32;
+ testDescription.mBytesPerFrame = testDescription.mBitsPerChannel/8 * testDescription.mChannelsPerFrame;
+ testDescription.mBytesPerPacket = testDescription.mBytesPerFrame * testDescription.mFramesPerPacket;
formatFlags = (description.mFormatFlags | kLinearPCMFormatFlagIsSignedInteger) & ~kLinearPCMFormatFlagIsFloat;
testDescription.mFormatFlags = formatFlags;
- result = AudioStreamSetProperty( streamIDs[firstStream+i], NULL, 0, kAudioStreamPropertyPhysicalFormat, dataSize, &testDescription );
+ result = AudioObjectSetPropertyData( streamIDs[firstStream+i], &property, 0, NULL, dataSize, &testDescription );
if ( result == noErr ) continue;
testDescription = description;
testDescription.mBitsPerChannel = 24;
+ testDescription.mBytesPerFrame = testDescription.mBitsPerChannel/8 * testDescription.mChannelsPerFrame;
+ testDescription.mBytesPerPacket = testDescription.mBytesPerFrame * testDescription.mFramesPerPacket;
testDescription.mFormatFlags = formatFlags;
- result = AudioStreamSetProperty( streamIDs[firstStream+i], NULL, 0, kAudioStreamPropertyPhysicalFormat, dataSize, &testDescription );
+ result = AudioObjectSetPropertyData( streamIDs[firstStream+i], &property, 0, NULL, dataSize, &testDescription );
if ( result == noErr ) continue;
testDescription = description;
testDescription.mBitsPerChannel = 16;
+ testDescription.mBytesPerFrame = testDescription.mBitsPerChannel/8 * testDescription.mChannelsPerFrame;
+ testDescription.mBytesPerPacket = testDescription.mBytesPerFrame * testDescription.mFramesPerPacket;
testDescription.mFormatFlags = formatFlags;
- result = AudioStreamSetProperty( streamIDs[firstStream+i], NULL, 0, kAudioStreamPropertyPhysicalFormat, dataSize, &testDescription );
+ result = AudioObjectSetPropertyData( streamIDs[firstStream+i], &property, 0, NULL, dataSize, &testDescription );
if ( result == noErr ) continue;
testDescription = description;
testDescription.mBitsPerChannel = 8;
+ testDescription.mBytesPerFrame = testDescription.mBitsPerChannel/8 * testDescription.mChannelsPerFrame;
+ testDescription.mBytesPerPacket = testDescription.mBytesPerFrame * testDescription.mFramesPerPacket;
testDescription.mFormatFlags = formatFlags;
- result = AudioStreamSetProperty( streamIDs[firstStream+i], NULL, 0, kAudioStreamPropertyPhysicalFormat, dataSize, &testDescription );
+ result = AudioObjectSetPropertyData( streamIDs[firstStream+i], &property, 0, NULL, dataSize, &testDescription );
if ( result != noErr ) {
errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") setting physical data format for device (" << device << ").";
errorText_ = errorStream_.str();
}
}
+
// Get the stream latency. There can be latency in both the device
// and the stream. First, attempt to get the device latency on the
// master channel or the first open channel. Errors that might
// occur here are not deemed critical.
- // ***** CHECK THIS ***** //
- UInt32 latency, channel = 0;
+ UInt32 latency;
dataSize = sizeof( UInt32 );
- AudioDevicePropertyID property = kAudioDevicePropertyLatency;
- if ( hasProperty( id, channel, isInput, property ) == true ) {
- result = AudioDeviceGetProperty( id, channel, isInput, property, &dataSize, &latency );
+ property.mSelector = kAudioDevicePropertyLatency;
+ if ( AudioObjectHasProperty( id, &property ) == true ) {
+ result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, &latency );
if ( result == kAudioHardwareNoError ) stream_.latency[ mode ] = latency;
else {
errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting device latency for device (" << device << ").";
// Now try to get the stream latency. For multiple streams, I assume the
// latency is equal for each.
- result = AudioStreamGetProperty( streamIDs[firstStream], 0, property, &dataSize, &latency );
+ result = AudioObjectGetPropertyData( streamIDs[firstStream], &property, 0, NULL, &dataSize, &latency );
if ( result == kAudioHardwareNoError ) stream_.latency[ mode ] += latency;
else {
errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting stream latency for device (" << device << ").";
else if ( monoMode && stream_.userInterleaved )
stream_.doConvertBuffer[mode] = true;
- std::cout << "doConvert = " << stream_.doConvertBuffer[mode] << ", userInterleaved = " << stream_.userInterleaved << ", deviceInterleaved = " << stream_.deviceInterleaved[mode] << std::endl;
-
// Allocate our CoreHandle structure for the stream.
CoreHandle *handle = 0;
if ( stream_.apiHandle == 0 ) {
}
// Setup the device property listener for over/underload.
- result = AudioDeviceAddPropertyListener( id, 0, isInput,
- kAudioDeviceProcessorOverload,
- deviceListener, (void *) handle );
+ property.mSelector = kAudioDeviceProcessorOverload;
+ result = AudioObjectAddPropertyListener( id, &property, deviceListener, (void *) handle );
return SUCCESS;
MUTEX_LOCK( &stream_.mutex );
+ if ( stream_.state == STREAM_STOPPED ) {
+ MUTEX_UNLOCK( &stream_.mutex );
+ return;
+ }
+
OSStatus result = noErr;
CoreHandle *handle = (CoreHandle *) stream_.apiHandle;
if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
pthread_cond_wait( &handle->condition, &stream_.mutex ); // block until signaled
}
+ MUTEX_UNLOCK( &stream_.mutex );
result = AudioDeviceStop( handle->id[0], callbackHandler );
+ MUTEX_LOCK( &stream_.mutex );
if ( result != noErr ) {
errorStream_ << "RtApiCore::stopStream: system error (" << getErrorCode( result ) << ") stopping callback procedure on device (" << stream_.device[0] << ").";
errorText_ = errorStream_.str();
}
}
+ stream_.state = STREAM_STOPPED;
+
unlock:
MUTEX_UNLOCK( &stream_.mutex );
- stream_.state = STREAM_STOPPED;
if ( result == noErr ) return;
error( RtError::SYSTEM_ERROR );
}
MUTEX_LOCK( &stream_.mutex );
+ // The state might change while waiting on a mutex.
+ if ( stream_.state == STREAM_STOPPED ) {
+ MUTEX_UNLOCK( &stream_.mutex );
+ return SUCCESS;
+ }
+
AudioDeviceID outputDevice = handle->id[0];
// Invoke user callback to get fresh output data UNLESS we are
UInt32 bufferBytes = outBufferList->mBuffers[handle->iStream[0]].mDataByteSize;
for ( unsigned int i=0; i<stream_.nUserChannels[0]; i++ ) {
memcpy( outBufferList->mBuffers[handle->iStream[0]+i].mData,
- &inBuffer[i*bufferBytes], bufferBytes );
+ (void *)&inBuffer[i*stream_.bufferSize], bufferBytes );
}
}
else { // fill multiple multi-channel streams with interleaved data
bool inInterleaved = ( stream_.userInterleaved ) ? true : false;
UInt32 inChannels = stream_.nUserChannels[0];
if ( stream_.doConvertBuffer[0] ) {
- inInterleaved = true; // device buffer will always be interleaved for nStreams > 1
+ inInterleaved = true; // device buffer will always be interleaved for nStreams > 1 and not mono mode
inChannels = stream_.nDeviceChannels[0];
}
if ( stream_.deviceInterleaved[1] == false ) { // mono mode
UInt32 bufferBytes = inBufferList->mBuffers[handle->iStream[1]].mDataByteSize;
for ( unsigned int i=0; i<stream_.nUserChannels[1]; i++ ) {
- memcpy( &outBuffer[i*bufferBytes],
+ memcpy( (void *)&outBuffer[i*stream_.bufferSize],
inBufferList->mBuffers[handle->iStream[1]+i].mData, bufferBytes );
}
}
bool outInterleaved = ( stream_.userInterleaved ) ? true : false;
UInt32 outChannels = stream_.nUserChannels[1];
if ( stream_.doConvertBuffer[1] ) {
- outInterleaved = true; // device buffer will always be interleaved for nStreams > 1
+ outInterleaved = true; // device buffer will always be interleaved for nStreams > 1 and not mono mode
outChannels = stream_.nDeviceChannels[1];
}
return SUCCESS;
}
- const char* RtApiCore :: getErrorCode( OSStatus code )
- {
- switch( code ) {
+const char* RtApiCore :: getErrorCode( OSStatus code )
+{
+ switch( code ) {
- case kAudioHardwareNotRunningError:
- return "kAudioHardwareNotRunningError";
+ case kAudioHardwareNotRunningError:
+ return "kAudioHardwareNotRunningError";
- case kAudioHardwareUnspecifiedError:
- return "kAudioHardwareUnspecifiedError";
+ case kAudioHardwareUnspecifiedError:
+ return "kAudioHardwareUnspecifiedError";
- case kAudioHardwareUnknownPropertyError:
- return "kAudioHardwareUnknownPropertyError";
+ case kAudioHardwareUnknownPropertyError:
+ return "kAudioHardwareUnknownPropertyError";
- case kAudioHardwareBadPropertySizeError:
- return "kAudioHardwareBadPropertySizeError";
+ case kAudioHardwareBadPropertySizeError:
+ return "kAudioHardwareBadPropertySizeError";
- case kAudioHardwareIllegalOperationError:
- return "kAudioHardwareIllegalOperationError";
+ case kAudioHardwareIllegalOperationError:
+ return "kAudioHardwareIllegalOperationError";
- case kAudioHardwareBadObjectError:
- return "kAudioHardwareBadObjectError";
+ case kAudioHardwareBadObjectError:
+ return "kAudioHardwareBadObjectError";
- case kAudioHardwareBadDeviceError:
- return "kAudioHardwareBadDeviceError";
+ case kAudioHardwareBadDeviceError:
+ return "kAudioHardwareBadDeviceError";
- case kAudioHardwareBadStreamError:
- return "kAudioHardwareBadStreamError";
+ case kAudioHardwareBadStreamError:
+ return "kAudioHardwareBadStreamError";
- case kAudioHardwareUnsupportedOperationError:
- return "kAudioHardwareUnsupportedOperationError";
+ case kAudioHardwareUnsupportedOperationError:
+ return "kAudioHardwareUnsupportedOperationError";
- case kAudioDeviceUnsupportedFormatError:
- return "kAudioDeviceUnsupportedFormatError";
+ case kAudioDeviceUnsupportedFormatError:
+ return "kAudioDeviceUnsupportedFormatError";
- case kAudioDevicePermissionsError:
- return "kAudioDevicePermissionsError";
+ case kAudioDevicePermissionsError:
+ return "kAudioDevicePermissionsError";
- default:
- return "CoreAudio unknown error";
- }
+ default:
+ return "CoreAudio unknown error";
}
+}
//******************** End of __MACOSX_CORE__ *********************//
#endif
#if defined(__UNIX_JACK__)
- // JACK is a low-latency audio server, originally written for the
- // GNU/Linux operating system and now also ported to OS-X. It can
- // connect a number of different applications to an audio device, as
- // well as allowing them to share audio between themselves.
- //
- // When using JACK with RtAudio, "devices" refer to JACK clients that
- // have ports connected to the server. The JACK server is typically
- // started in a terminal as follows:
- //
- // .jackd -d alsa -d hw:0
- //
- // or through an interface program such as qjackctl. Many of the
- // parameters normally set for a stream are fixed by the JACK server
- // and can be specified when the JACK server is started. In
- // particular,
- //
- // .jackd -d alsa -d hw:0 -r 44100 -p 512 -n 4
- //
- // specifies a sample rate of 44100 Hz, a buffer size of 512 sample
- // frames, and number of buffers = 4. Once the server is running, it
- // is not possible to override these values. If the values are not
- // specified in the command-line, the JACK server uses default values.
- //
- // The JACK server does not have to be running when an instance of
- // RtApiJack is created, though the function getDeviceCount() will
- // report 0 devices found until JACK has been started. When no
- // devices are available (i.e., the JACK server is not running), a
- // stream cannot be opened.
+// JACK is a low-latency audio server, originally written for the
+// GNU/Linux operating system and now also ported to OS-X. It can
+// connect a number of different applications to an audio device, as
+// well as allowing them to share audio between themselves.
+//
+// When using JACK with RtAudio, "devices" refer to JACK clients that
+// have ports connected to the server. The JACK server is typically
+// started in a terminal as follows:
+//
+// .jackd -d alsa -d hw:0
+//
+// or through an interface program such as qjackctl. Many of the
+// parameters normally set for a stream are fixed by the JACK server
+// and can be specified when the JACK server is started. In
+// particular,
+//
+// .jackd -d alsa -d hw:0 -r 44100 -p 512 -n 4
+//
+// specifies a sample rate of 44100 Hz, a buffer size of 512 sample
+// frames, and number of buffers = 4. Once the server is running, it
+// is not possible to override these values. If the values are not
+// specified in the command-line, the JACK server uses default values.
+//
+// The JACK server does not have to be running when an instance of
+// RtApiJack is created, though the function getDeviceCount() will
+// report 0 devices found until JACK has been started. When no
+// devices are available (i.e., the JACK server is not running), a
+// stream cannot be opened.
#include <jack/jack.h>
#include <unistd.h>
+#include <cstdio>
- // A structure to hold various information related to the Jack API
- // implementation.
- struct JackHandle {
- jack_client_t *client;
- jack_port_t **ports[2];
- std::string deviceName[2];
- bool xrun[2];
- pthread_cond_t condition;
- int drainCounter; // Tracks callback counts when draining
- bool internalDrain; // Indicates if stop is initiated from callback or not.
-
- JackHandle()
- :client(0), drainCounter(0), internalDrain(false) { ports[0] = 0; ports[1] = 0; xrun[0] = false; xrun[1] = false; }
- };
-
- RtApiJack :: RtApiJack()
- {
- // Nothing to do here.
- }
-
- RtApiJack :: ~RtApiJack()
- {
- if ( stream_.state != STREAM_CLOSED ) closeStream();
- }
+// A structure to hold various information related to the Jack API
+// implementation.
+struct JackHandle {
+ jack_client_t *client;
+ jack_port_t **ports[2];
+ std::string deviceName[2];
+ bool xrun[2];
+ pthread_cond_t condition;
+ int drainCounter; // Tracks callback counts when draining
+ bool internalDrain; // Indicates if stop is initiated from callback or not.
- unsigned int RtApiJack :: getDeviceCount( void )
- {
- // See if we can become a jack client.
- jack_options_t options = (jack_options_t) ( JackNoStartServer | JackUseExactName ); //JackNullOption;
- jack_status_t *status = NULL;
- jack_client_t *client = jack_client_open( "RtApiJackCount", options, status );
- if ( client == 0 ) return 0;
-
- const char **ports;
- std::string port, previousPort;
- unsigned int nChannels = 0, nDevices = 0;
- ports = jack_get_ports( client, NULL, NULL, 0 );
- if ( ports ) {
- // Parse the port names up to the first colon (:).
- size_t iColon = 0;
- do {
- port = (char *) ports[ nChannels ];
- iColon = port.find(":");
- if ( iColon != std::string::npos ) {
- port = port.substr( 0, iColon + 1 );
- if ( port != previousPort ) {
- nDevices++;
- previousPort = port;
- }
- }
- } while ( ports[++nChannels] );
- free( ports );
- }
+ JackHandle()
+ :client(0), drainCounter(0), internalDrain(false) { ports[0] = 0; ports[1] = 0; xrun[0] = false; xrun[1] = false; }
+};
- jack_client_close( client );
- return nDevices;
- }
+ThreadHandle threadId;
+void jackSilentError( const char * ) {};
- RtAudio::DeviceInfo RtApiJack :: getDeviceInfo( unsigned int device )
- {
- RtAudio::DeviceInfo info;
- info.probed = false;
+RtApiJack :: RtApiJack()
+{
+ // Nothing to do here.
+#if !defined(__RTAUDIO_DEBUG__)
+ // Turn off Jack's internal error reporting.
+ jack_set_error_function( &jackSilentError );
+#endif
+}
- jack_options_t options = (jack_options_t) ( JackNoStartServer | JackUseExactName ); //JackNullOption
- jack_status_t *status = NULL;
- jack_client_t *client = jack_client_open( "RtApiJackInfo", options, status );
- if ( client == 0 ) {
- errorText_ = "RtApiJack::getDeviceInfo: Jack server not found or connection error!";
- error( RtError::WARNING );
- return info;
- }
+RtApiJack :: ~RtApiJack()
+{
+ if ( stream_.state != STREAM_CLOSED ) closeStream();
+}
- const char **ports;
- std::string port, previousPort;
- unsigned int nPorts = 0, nDevices = 0;
- ports = jack_get_ports( client, NULL, NULL, 0 );
- if ( ports ) {
- // Parse the port names up to the first colon (:).
- size_t iColon = 0;
- do {
- port = (char *) ports[ nPorts ];
- iColon = port.find(":");
- if ( iColon != std::string::npos ) {
- port = port.substr( 0, iColon );
- if ( port != previousPort ) {
- if ( nDevices == device ) info.name = port;
- nDevices++;
- previousPort = port;
- }
+unsigned int RtApiJack :: getDeviceCount( void )
+{
+ // See if we can become a jack client.
+ jack_options_t options = (jack_options_t) ( JackNoStartServer | JackUseExactName ); //JackNullOption;
+ jack_status_t *status = NULL;
+ jack_client_t *client = jack_client_open( "RtApiJackCount", options, status );
+ if ( client == 0 ) return 0;
+
+ const char **ports;
+ std::string port, previousPort;
+ unsigned int nChannels = 0, nDevices = 0;
+ ports = jack_get_ports( client, NULL, NULL, 0 );
+ if ( ports ) {
+ // Parse the port names up to the first colon (:).
+ size_t iColon = 0;
+ do {
+ port = (char *) ports[ nChannels ];
+ iColon = port.find(":");
+ if ( iColon != std::string::npos ) {
+ port = port.substr( 0, iColon + 1 );
+ if ( port != previousPort ) {
+ nDevices++;
+ previousPort = port;
}
- } while ( ports[++nPorts] );
- free( ports );
- }
-
- if ( device >= nDevices ) {
- errorText_ = "RtApiJack::getDeviceInfo: device ID is invalid!";
- error( RtError::INVALID_USE );
- }
+ }
+ } while ( ports[++nChannels] );
+ free( ports );
+ }
- // Get the current jack server sample rate.
- info.sampleRates.clear();
- info.sampleRates.push_back( jack_get_sample_rate( client ) );
+ jack_client_close( client );
+ return nDevices;
+}
- // Count the available ports containing the client name as device
- // channels. Jack "input ports" equal RtAudio output channels.
- unsigned int nChannels = 0;
- ports = jack_get_ports( client, info.name.c_str(), NULL, JackPortIsInput );
- if ( ports ) {
- while ( ports[ nChannels ] ) nChannels++;
- free( ports );
- info.outputChannels = nChannels;
- }
+RtAudio::DeviceInfo RtApiJack :: getDeviceInfo( unsigned int device )
+{
+ RtAudio::DeviceInfo info;
+ info.probed = false;
- // Jack "output ports" equal RtAudio input channels.
- nChannels = 0;
- ports = jack_get_ports( client, info.name.c_str(), NULL, JackPortIsOutput );
- if ( ports ) {
- while ( ports[ nChannels ] ) nChannels++;
- free( ports );
- info.inputChannels = nChannels;
- }
+ jack_options_t options = (jack_options_t) ( JackNoStartServer | JackUseExactName ); //JackNullOption
+ jack_status_t *status = NULL;
+ jack_client_t *client = jack_client_open( "RtApiJackInfo", options, status );
+ if ( client == 0 ) {
+ errorText_ = "RtApiJack::getDeviceInfo: Jack server not found or connection error!";
+ error( RtError::WARNING );
+ return info;
+ }
- if ( info.outputChannels == 0 && info.inputChannels == 0 ) {
- jack_client_close(client);
- errorText_ = "RtApiJack::getDeviceInfo: error determining Jack input/output channels!";
- error( RtError::WARNING );
- return info;
- }
+ const char **ports;
+ std::string port, previousPort;
+ unsigned int nPorts = 0, nDevices = 0;
+ ports = jack_get_ports( client, NULL, NULL, 0 );
+ if ( ports ) {
+ // Parse the port names up to the first colon (:).
+ size_t iColon = 0;
+ do {
+ port = (char *) ports[ nPorts ];
+ iColon = port.find(":");
+ if ( iColon != std::string::npos ) {
+ port = port.substr( 0, iColon );
+ if ( port != previousPort ) {
+ if ( nDevices == device ) info.name = port;
+ nDevices++;
+ previousPort = port;
+ }
+ }
+ } while ( ports[++nPorts] );
+ free( ports );
+ }
- // If device opens for both playback and capture, we determine the channels.
- if ( info.outputChannels > 0 && info.inputChannels > 0 )
- info.duplexChannels = (info.outputChannels > info.inputChannels) ? info.inputChannels : info.outputChannels;
+ if ( device >= nDevices ) {
+ errorText_ = "RtApiJack::getDeviceInfo: device ID is invalid!";
+ error( RtError::INVALID_USE );
+ }
- // Jack always uses 32-bit floats.
- info.nativeFormats = RTAUDIO_FLOAT32;
+ // Get the current jack server sample rate.
+ info.sampleRates.clear();
+ info.sampleRates.push_back( jack_get_sample_rate( client ) );
+
+ // Count the available ports containing the client name as device
+ // channels. Jack "input ports" equal RtAudio output channels.
+ unsigned int nChannels = 0;
+ ports = jack_get_ports( client, info.name.c_str(), NULL, JackPortIsInput );
+ if ( ports ) {
+ while ( ports[ nChannels ] ) nChannels++;
+ free( ports );
+ info.outputChannels = nChannels;
+ }
- // Jack doesn't provide default devices so we'll use the first available one.
- if ( device == 0 && info.outputChannels > 0 )
- info.isDefaultOutput = true;
- if ( device == 0 && info.inputChannels > 0 )
- info.isDefaultInput = true;
+ // Jack "output ports" equal RtAudio input channels.
+ nChannels = 0;
+ ports = jack_get_ports( client, info.name.c_str(), NULL, JackPortIsOutput );
+ if ( ports ) {
+ while ( ports[ nChannels ] ) nChannels++;
+ free( ports );
+ info.inputChannels = nChannels;
+ }
+ if ( info.outputChannels == 0 && info.inputChannels == 0 ) {
jack_client_close(client);
- info.probed = true;
+ errorText_ = "RtApiJack::getDeviceInfo: error determining Jack input/output channels!";
+ error( RtError::WARNING );
return info;
}
- int jackCallbackHandler( jack_nframes_t nframes, void *infoPointer )
- {
- CallbackInfo *info = (CallbackInfo *) infoPointer;
+ // If device opens for both playback and capture, we determine the channels.
+ if ( info.outputChannels > 0 && info.inputChannels > 0 )
+ info.duplexChannels = (info.outputChannels > info.inputChannels) ? info.inputChannels : info.outputChannels;
- RtApiJack *object = (RtApiJack *) info->object;
- if ( object->callbackEvent( (unsigned long) nframes ) == false ) return 1;
+ // Jack always uses 32-bit floats.
+ info.nativeFormats = RTAUDIO_FLOAT32;
- return 0;
- }
+ // Jack doesn't provide default devices so we'll use the first available one.
+ if ( device == 0 && info.outputChannels > 0 )
+ info.isDefaultOutput = true;
+ if ( device == 0 && info.inputChannels > 0 )
+ info.isDefaultInput = true;
- void jackShutdown( void *infoPointer )
- {
- CallbackInfo *info = (CallbackInfo *) infoPointer;
- RtApiJack *object = (RtApiJack *) info->object;
+ jack_client_close(client);
+ info.probed = true;
+ return info;
+}
- // Check current stream state. If stopped, then we'll assume this
- // was called as a result of a call to RtApiJack::stopStream (the
- // deactivation of a client handle causes this function to be called).
- // If not, we'll assume the Jack server is shutting down or some
- // other problem occurred and we should close the stream.
- if ( object->isStreamRunning() == false ) return;
+int jackCallbackHandler( jack_nframes_t nframes, void *infoPointer )
+{
+ CallbackInfo *info = (CallbackInfo *) infoPointer;
- object->closeStream();
- std::cerr << "\nRtApiJack: the Jack server is shutting down this client ... stream stopped and closed!!\n" << std::endl;
- }
+ RtApiJack *object = (RtApiJack *) info->object;
+ if ( object->callbackEvent( (unsigned long) nframes ) == false ) return 1;
- int jackXrun( void *infoPointer )
- {
- JackHandle *handle = (JackHandle *) infoPointer;
+ return 0;
+}
- if ( handle->ports[0] ) handle->xrun[0] = true;
- if ( handle->ports[1] ) handle->xrun[1] = true;
+// This function will be called by a spawned thread when the Jack
+// server signals that it is shutting down. It is necessary to handle
+// it this way because the jackShutdown() function must return before
+// the jack_deactivate() function (in closeStream()) will return.
+extern "C" void *jackCloseStream( void *ptr )
+{
+ CallbackInfo *info = (CallbackInfo *) ptr;
+ RtApiJack *object = (RtApiJack *) info->object;
- return 0;
- }
+ object->closeStream();
- bool RtApiJack :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,
- unsigned int firstChannel, unsigned int sampleRate,
- RtAudioFormat format, unsigned int *bufferSize,
- RtAudio::StreamOptions *options )
- {
- JackHandle *handle = (JackHandle *) stream_.apiHandle;
-
- // Look for jack server and try to become a client (only do once per stream).
- jack_client_t *client = 0;
- if ( mode == OUTPUT || ( mode == INPUT && stream_.mode != OUTPUT ) ) {
- jack_options_t jackoptions = (jack_options_t) ( JackNoStartServer | JackUseExactName ); //JackNullOption;
- jack_status_t *status = NULL;
- if ( options && !options->streamName.empty() )
- client = jack_client_open( options->streamName.c_str(), jackoptions, status );
- else
- client = jack_client_open( "RtApiJack", jackoptions, status );
- if ( client == 0 ) {
- errorText_ = "RtApiJack::probeDeviceOpen: Jack server not found or connection error!";
- error( RtError::WARNING );
- return FAILURE;
- }
- }
- else {
- // The handle must have been created on an earlier pass.
- client = handle->client;
- }
-
- const char **ports;
- std::string port, previousPort, deviceName;
- unsigned int nPorts = 0, nDevices = 0;
- ports = jack_get_ports( client, NULL, NULL, 0 );
- if ( ports ) {
- // Parse the port names up to the first colon (:).
- size_t iColon = 0;
- do {
- port = (char *) ports[ nPorts ];
- iColon = port.find(":");
- if ( iColon != std::string::npos ) {
- port = port.substr( 0, iColon );
- if ( port != previousPort ) {
- if ( nDevices == device ) deviceName = port;
- nDevices++;
- previousPort = port;
- }
- }
- } while ( ports[++nPorts] );
- free( ports );
- }
+ pthread_exit( NULL );
+}
+void jackShutdown( void *infoPointer )
+{
+ CallbackInfo *info = (CallbackInfo *) infoPointer;
+ RtApiJack *object = (RtApiJack *) info->object;
- if ( device >= nDevices ) {
- errorText_ = "RtApiJack::probeDeviceOpen: device ID is invalid!";
- return FAILURE;
- }
+ // Check current stream state. If stopped, then we'll assume this
+ // was called as a result of a call to RtApiJack::stopStream (the
+ // deactivation of a client handle causes this function to be called).
+ // If not, we'll assume the Jack server is shutting down or some
+ // other problem occurred and we should close the stream.
+ if ( object->isStreamRunning() == false ) return;
- // Count the available ports containing the client name as device
- // channels. Jack "input ports" equal RtAudio output channels.
- unsigned int nChannels = 0;
- unsigned long flag = JackPortIsInput;
- if ( mode == INPUT ) flag = JackPortIsOutput;
- ports = jack_get_ports( client, deviceName.c_str(), NULL, flag );
- if ( ports ) {
- while ( ports[ nChannels ] ) nChannels++;
- free( ports );
- }
+ pthread_create( &threadId, NULL, jackCloseStream, info );
+ std::cerr << "\nRtApiJack: the Jack server is shutting down this client ... stream stopped and closed!!\n" << std::endl;
+}
- // Compare the jack ports for specified client to the requested number of channels.
- if ( nChannels < (channels + firstChannel) ) {
- errorStream_ << "RtApiJack::probeDeviceOpen: requested number of channels (" << channels << ") + offset (" << firstChannel << ") not found for specified device (" << device << ":" << deviceName << ").";
- errorText_ = errorStream_.str();
- return FAILURE;
- }
+int jackXrun( void *infoPointer )
+{
+ JackHandle *handle = (JackHandle *) infoPointer;
- // Check the jack server sample rate.
- unsigned int jackRate = jack_get_sample_rate( client );
- if ( sampleRate != jackRate ) {
- jack_client_close( client );
- errorStream_ << "RtApiJack::probeDeviceOpen: the requested sample rate (" << sampleRate << ") is different than the JACK server rate (" << jackRate << ").";
- errorText_ = errorStream_.str();
+ if ( handle->ports[0] ) handle->xrun[0] = true;
+ if ( handle->ports[1] ) handle->xrun[1] = true;
+
+ return 0;
+}
+
+bool RtApiJack :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,
+ unsigned int firstChannel, unsigned int sampleRate,
+ RtAudioFormat format, unsigned int *bufferSize,
+ RtAudio::StreamOptions *options )
+{
+ JackHandle *handle = (JackHandle *) stream_.apiHandle;
+
+ // Look for jack server and try to become a client (only do once per stream).
+ jack_client_t *client = 0;
+ if ( mode == OUTPUT || ( mode == INPUT && stream_.mode != OUTPUT ) ) {
+ jack_options_t jackoptions = (jack_options_t) ( JackNoStartServer | JackUseExactName ); //JackNullOption;
+ jack_status_t *status = NULL;
+ if ( options && !options->streamName.empty() )
+ client = jack_client_open( options->streamName.c_str(), jackoptions, status );
+ else
+ client = jack_client_open( "RtApiJack", jackoptions, status );
+ if ( client == 0 ) {
+ errorText_ = "RtApiJack::probeDeviceOpen: Jack server not found or connection error!";
+ error( RtError::WARNING );
return FAILURE;
}
- stream_.sampleRate = jackRate;
+ }
+ else {
+ // The handle must have been created on an earlier pass.
+ client = handle->client;
+ }
+
+ const char **ports;
+ std::string port, previousPort, deviceName;
+ unsigned int nPorts = 0, nDevices = 0;
+ ports = jack_get_ports( client, NULL, NULL, 0 );
+ if ( ports ) {
+ // Parse the port names up to the first colon (:).
+ size_t iColon = 0;
+ do {
+ port = (char *) ports[ nPorts ];
+ iColon = port.find(":");
+ if ( iColon != std::string::npos ) {
+ port = port.substr( 0, iColon );
+ if ( port != previousPort ) {
+ if ( nDevices == device ) deviceName = port;
+ nDevices++;
+ previousPort = port;
+ }
+ }
+ } while ( ports[++nPorts] );
+ free( ports );
+ }
+
+ if ( device >= nDevices ) {
+ errorText_ = "RtApiJack::probeDeviceOpen: device ID is invalid!";
+ return FAILURE;
+ }
- // Get the latency of the JACK port.
- ports = jack_get_ports( client, deviceName.c_str(), NULL, flag );
- if ( ports[ firstChannel ] )
- stream_.latency[mode] = jack_port_get_latency( jack_port_by_name( client, ports[ firstChannel ] ) );
+ // Count the available ports containing the client name as device
+ // channels. Jack "input ports" equal RtAudio output channels.
+ unsigned int nChannels = 0;
+ unsigned long flag = JackPortIsInput;
+ if ( mode == INPUT ) flag = JackPortIsOutput;
+ ports = jack_get_ports( client, deviceName.c_str(), NULL, flag );
+ if ( ports ) {
+ while ( ports[ nChannels ] ) nChannels++;
free( ports );
+ }
- // The jack server always uses 32-bit floating-point data.
- stream_.deviceFormat[mode] = RTAUDIO_FLOAT32;
- stream_.userFormat = format;
+ // Compare the jack ports for specified client to the requested number of channels.
+ if ( nChannels < (channels + firstChannel) ) {
+ errorStream_ << "RtApiJack::probeDeviceOpen: requested number of channels (" << channels << ") + offset (" << firstChannel << ") not found for specified device (" << device << ":" << deviceName << ").";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
- if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) stream_.userInterleaved = false;
- else stream_.userInterleaved = true;
+ // Check the jack server sample rate.
+ unsigned int jackRate = jack_get_sample_rate( client );
+ if ( sampleRate != jackRate ) {
+ jack_client_close( client );
+ errorStream_ << "RtApiJack::probeDeviceOpen: the requested sample rate (" << sampleRate << ") is different than the JACK server rate (" << jackRate << ").";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
+ stream_.sampleRate = jackRate;
- // Jack always uses non-interleaved buffers.
- stream_.deviceInterleaved[mode] = false;
+ // Get the latency of the JACK port.
+ ports = jack_get_ports( client, deviceName.c_str(), NULL, flag );
+ if ( ports[ firstChannel ] )
+ stream_.latency[mode] = jack_port_get_latency( jack_port_by_name( client, ports[ firstChannel ] ) );
+ free( ports );
- // Jack always provides host byte-ordered data.
- stream_.doByteSwap[mode] = false;
+ // The jack server always uses 32-bit floating-point data.
+ stream_.deviceFormat[mode] = RTAUDIO_FLOAT32;
+ stream_.userFormat = format;
- // Get the buffer size. The buffer size and number of buffers
- // (periods) is set when the jack server is started.
- stream_.bufferSize = (int) jack_get_buffer_size( client );
- *bufferSize = stream_.bufferSize;
+ if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) stream_.userInterleaved = false;
+ else stream_.userInterleaved = true;
- stream_.nDeviceChannels[mode] = channels;
- stream_.nUserChannels[mode] = channels;
+ // Jack always uses non-interleaved buffers.
+ stream_.deviceInterleaved[mode] = false;
- // Set flags for buffer conversion.
- stream_.doConvertBuffer[mode] = false;
- if ( stream_.userFormat != stream_.deviceFormat[mode] )
- stream_.doConvertBuffer[mode] = true;
- if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] &&
- stream_.nUserChannels[mode] > 1 )
- stream_.doConvertBuffer[mode] = true;
+ // Jack always provides host byte-ordered data.
+ stream_.doByteSwap[mode] = false;
- // Allocate our JackHandle structure for the stream.
- if ( handle == 0 ) {
- try {
- handle = new JackHandle;
- }
- catch ( std::bad_alloc& ) {
- errorText_ = "RtApiJack::probeDeviceOpen: error allocating JackHandle memory.";
- goto error;
- }
+ // Get the buffer size. The buffer size and number of buffers
+ // (periods) is set when the jack server is started.
+ stream_.bufferSize = (int) jack_get_buffer_size( client );
+ *bufferSize = stream_.bufferSize;
- if ( pthread_cond_init(&handle->condition, NULL) ) {
- errorText_ = "RtApiJack::probeDeviceOpen: error initializing pthread condition variable.";
- goto error;
- }
- stream_.apiHandle = (void *) handle;
- handle->client = client;
+ stream_.nDeviceChannels[mode] = channels;
+ stream_.nUserChannels[mode] = channels;
+
+ // Set flags for buffer conversion.
+ stream_.doConvertBuffer[mode] = false;
+ if ( stream_.userFormat != stream_.deviceFormat[mode] )
+ stream_.doConvertBuffer[mode] = true;
+ if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] &&
+ stream_.nUserChannels[mode] > 1 )
+ stream_.doConvertBuffer[mode] = true;
+
+ // Allocate our JackHandle structure for the stream.
+ if ( handle == 0 ) {
+ try {
+ handle = new JackHandle;
+ }
+ catch ( std::bad_alloc& ) {
+ errorText_ = "RtApiJack::probeDeviceOpen: error allocating JackHandle memory.";
+ goto error;
}
- handle->deviceName[mode] = deviceName;
- // Allocate necessary internal buffers.
- unsigned long bufferBytes;
- bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat );
- stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 );
- if ( stream_.userBuffer[mode] == NULL ) {
- errorText_ = "RtApiJack::probeDeviceOpen: error allocating user buffer memory.";
+ if ( pthread_cond_init(&handle->condition, NULL) ) {
+ errorText_ = "RtApiJack::probeDeviceOpen: error initializing pthread condition variable.";
goto error;
}
+ stream_.apiHandle = (void *) handle;
+ handle->client = client;
+ }
+ handle->deviceName[mode] = deviceName;
- if ( stream_.doConvertBuffer[mode] ) {
+ // Allocate necessary internal buffers.
+ unsigned long bufferBytes;
+ bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat );
+ stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 );
+ if ( stream_.userBuffer[mode] == NULL ) {
+ errorText_ = "RtApiJack::probeDeviceOpen: error allocating user buffer memory.";
+ goto error;
+ }
- bool makeBuffer = true;
- if ( mode == OUTPUT )
- bufferBytes = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] );
- else { // mode == INPUT
- bufferBytes = stream_.nDeviceChannels[1] * formatBytes( stream_.deviceFormat[1] );
- if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) {
- unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes(stream_.deviceFormat[0]);
- if ( bufferBytes < bytesOut ) makeBuffer = false;
- }
- }
+ if ( stream_.doConvertBuffer[mode] ) {
- if ( makeBuffer ) {
- bufferBytes *= *bufferSize;
- if ( stream_.deviceBuffer ) free( stream_.deviceBuffer );
- stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 );
- if ( stream_.deviceBuffer == NULL ) {
- errorText_ = "RtApiJack::probeDeviceOpen: error allocating device buffer memory.";
- goto error;
- }
+ bool makeBuffer = true;
+ if ( mode == OUTPUT )
+ bufferBytes = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] );
+ else { // mode == INPUT
+ bufferBytes = stream_.nDeviceChannels[1] * formatBytes( stream_.deviceFormat[1] );
+ if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) {
+ unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes(stream_.deviceFormat[0]);
+ if ( bufferBytes < bytesOut ) makeBuffer = false;
}
}
- // Allocate memory for the Jack ports (channels) identifiers.
- handle->ports[mode] = (jack_port_t **) malloc ( sizeof (jack_port_t *) * channels );
- if ( handle->ports[mode] == NULL ) {
- errorText_ = "RtApiJack::probeDeviceOpen: error allocating port memory.";
- goto error;
+ if ( makeBuffer ) {
+ bufferBytes *= *bufferSize;
+ if ( stream_.deviceBuffer ) free( stream_.deviceBuffer );
+ stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 );
+ if ( stream_.deviceBuffer == NULL ) {
+ errorText_ = "RtApiJack::probeDeviceOpen: error allocating device buffer memory.";
+ goto error;
+ }
}
+ }
- stream_.device[mode] = device;
- stream_.channelOffset[mode] = firstChannel;
- stream_.state = STREAM_STOPPED;
- stream_.callbackInfo.object = (void *) this;
+ // Allocate memory for the Jack ports (channels) identifiers.
+ handle->ports[mode] = (jack_port_t **) malloc ( sizeof (jack_port_t *) * channels );
+ if ( handle->ports[mode] == NULL ) {
+ errorText_ = "RtApiJack::probeDeviceOpen: error allocating port memory.";
+ goto error;
+ }
- if ( stream_.mode == OUTPUT && mode == INPUT )
- // We had already set up the stream for output.
- stream_.mode = DUPLEX;
- else {
- stream_.mode = mode;
- jack_set_process_callback( handle->client, jackCallbackHandler, (void *) &stream_.callbackInfo );
- jack_set_xrun_callback( handle->client, jackXrun, (void *) &handle );
- jack_on_shutdown( handle->client, jackShutdown, (void *) &stream_.callbackInfo );
- }
+ stream_.device[mode] = device;
+ stream_.channelOffset[mode] = firstChannel;
+ stream_.state = STREAM_STOPPED;
+ stream_.callbackInfo.object = (void *) this;
- // Register our ports.
- char label[64];
- if ( mode == OUTPUT ) {
- for ( unsigned int i=0; i<stream_.nUserChannels[0]; i++ ) {
- snprintf( label, 64, "outport %d", i );
- handle->ports[0][i] = jack_port_register( handle->client, (const char *)label,
- JACK_DEFAULT_AUDIO_TYPE, JackPortIsOutput, 0 );
- }
+ if ( stream_.mode == OUTPUT && mode == INPUT )
+ // We had already set up the stream for output.
+ stream_.mode = DUPLEX;
+ else {
+ stream_.mode = mode;
+ jack_set_process_callback( handle->client, jackCallbackHandler, (void *) &stream_.callbackInfo );
+ jack_set_xrun_callback( handle->client, jackXrun, (void *) &handle );
+ jack_on_shutdown( handle->client, jackShutdown, (void *) &stream_.callbackInfo );
+ }
+
+ // Register our ports.
+ char label[64];
+ if ( mode == OUTPUT ) {
+ for ( unsigned int i=0; i<stream_.nUserChannels[0]; i++ ) {
+ snprintf( label, 64, "outport %d", i );
+ handle->ports[0][i] = jack_port_register( handle->client, (const char *)label,
+ JACK_DEFAULT_AUDIO_TYPE, JackPortIsOutput, 0 );
}
- else {
- for ( unsigned int i=0; i<stream_.nUserChannels[1]; i++ ) {
- snprintf( label, 64, "inport %d", i );
- handle->ports[1][i] = jack_port_register( handle->client, (const char *)label,
- JACK_DEFAULT_AUDIO_TYPE, JackPortIsInput, 0 );
- }
+ }
+ else {
+ for ( unsigned int i=0; i<stream_.nUserChannels[1]; i++ ) {
+ snprintf( label, 64, "inport %d", i );
+ handle->ports[1][i] = jack_port_register( handle->client, (const char *)label,
+ JACK_DEFAULT_AUDIO_TYPE, JackPortIsInput, 0 );
}
+ }
- // Setup the buffer conversion information structure. We don't use
- // buffers to do channel offsets, so we override that parameter
- // here.
- if ( stream_.doConvertBuffer[mode] ) setConvertInfo( mode, 0 );
+ // Setup the buffer conversion information structure. We don't use
+ // buffers to do channel offsets, so we override that parameter
+ // here.
+ if ( stream_.doConvertBuffer[mode] ) setConvertInfo( mode, 0 );
- return SUCCESS;
+ return SUCCESS;
- error:
- if ( handle ) {
- pthread_cond_destroy( &handle->condition );
- jack_client_close( handle->client );
+ error:
+ if ( handle ) {
+ pthread_cond_destroy( &handle->condition );
+ jack_client_close( handle->client );
- if ( handle->ports[0] ) free( handle->ports[0] );
- if ( handle->ports[1] ) free( handle->ports[1] );
+ if ( handle->ports[0] ) free( handle->ports[0] );
+ if ( handle->ports[1] ) free( handle->ports[1] );
- delete handle;
- stream_.apiHandle = 0;
- }
+ delete handle;
+ stream_.apiHandle = 0;
+ }
- for ( int i=0; i<2; i++ ) {
- if ( stream_.userBuffer[i] ) {
- free( stream_.userBuffer[i] );
- stream_.userBuffer[i] = 0;
- }
+ for ( int i=0; i<2; i++ ) {
+ if ( stream_.userBuffer[i] ) {
+ free( stream_.userBuffer[i] );
+ stream_.userBuffer[i] = 0;
}
+ }
- if ( stream_.deviceBuffer ) {
- free( stream_.deviceBuffer );
- stream_.deviceBuffer = 0;
- }
+ if ( stream_.deviceBuffer ) {
+ free( stream_.deviceBuffer );
+ stream_.deviceBuffer = 0;
+ }
- return FAILURE;
+ return FAILURE;
+}
+
+void RtApiJack :: closeStream( void )
+{
+ if ( stream_.state == STREAM_CLOSED ) {
+ errorText_ = "RtApiJack::closeStream(): no open stream to close!";
+ error( RtError::WARNING );
+ return;
}
- void RtApiJack :: closeStream( void )
- {
- if ( stream_.state == STREAM_CLOSED ) {
- errorText_ = "RtApiJack::closeStream(): no open stream to close!";
- error( RtError::WARNING );
- return;
- }
+ JackHandle *handle = (JackHandle *) stream_.apiHandle;
+ if ( handle ) {
- JackHandle *handle = (JackHandle *) stream_.apiHandle;
- if ( handle ) {
+ if ( stream_.state == STREAM_RUNNING )
+ jack_deactivate( handle->client );
- if ( stream_.state == STREAM_RUNNING )
- jack_deactivate( handle->client );
+ jack_client_close( handle->client );
+ }
- jack_client_close( handle->client );
- }
+ if ( handle ) {
+ if ( handle->ports[0] ) free( handle->ports[0] );
+ if ( handle->ports[1] ) free( handle->ports[1] );
+ pthread_cond_destroy( &handle->condition );
+ delete handle;
+ stream_.apiHandle = 0;
+ }
- if ( handle ) {
- if ( handle->ports[0] ) free( handle->ports[0] );
- if ( handle->ports[1] ) free( handle->ports[1] );
- pthread_cond_destroy( &handle->condition );
- delete handle;
- stream_.apiHandle = 0;
+ for ( int i=0; i<2; i++ ) {
+ if ( stream_.userBuffer[i] ) {
+ free( stream_.userBuffer[i] );
+ stream_.userBuffer[i] = 0;
}
+ }
- for ( int i=0; i<2; i++ ) {
- if ( stream_.userBuffer[i] ) {
- free( stream_.userBuffer[i] );
- stream_.userBuffer[i] = 0;
- }
- }
+ if ( stream_.deviceBuffer ) {
+ free( stream_.deviceBuffer );
+ stream_.deviceBuffer = 0;
+ }
- if ( stream_.deviceBuffer ) {
- free( stream_.deviceBuffer );
- stream_.deviceBuffer = 0;
- }
+ stream_.mode = UNINITIALIZED;
+ stream_.state = STREAM_CLOSED;
+}
- stream_.mode = UNINITIALIZED;
- stream_.state = STREAM_CLOSED;
+void RtApiJack :: startStream( void )
+{
+ verifyStream();
+ if ( stream_.state == STREAM_RUNNING ) {
+ errorText_ = "RtApiJack::startStream(): the stream is already running!";
+ error( RtError::WARNING );
+ return;
}
- void RtApiJack :: startStream( void )
- {
- verifyStream();
- if ( stream_.state == STREAM_RUNNING ) {
- errorText_ = "RtApiJack::startStream(): the stream is already running!";
- error( RtError::WARNING );
- return;
- }
+ MUTEX_LOCK(&stream_.mutex);
- MUTEX_LOCK(&stream_.mutex);
+ JackHandle *handle = (JackHandle *) stream_.apiHandle;
+ int result = jack_activate( handle->client );
+ if ( result ) {
+ errorText_ = "RtApiJack::startStream(): unable to activate JACK client!";
+ goto unlock;
+ }
- JackHandle *handle = (JackHandle *) stream_.apiHandle;
- int result = jack_activate( handle->client );
- if ( result ) {
- errorText_ = "RtApiJack::startStream(): unable to activate JACK client!";
+ const char **ports;
+
+ // Get the list of available ports.
+ if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
+ result = 1;
+ ports = jack_get_ports( handle->client, handle->deviceName[0].c_str(), NULL, JackPortIsInput);
+ if ( ports == NULL) {
+ errorText_ = "RtApiJack::startStream(): error determining available JACK input ports!";
goto unlock;
}
- const char **ports;
-
- // Get the list of available ports.
- if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
+ // Now make the port connections. Since RtAudio wasn't designed to
+ // allow the user to select particular channels of a device, we'll
+ // just open the first "nChannels" ports with offset.
+ for ( unsigned int i=0; i<stream_.nUserChannels[0]; i++ ) {
result = 1;
- ports = jack_get_ports( handle->client, handle->deviceName[0].c_str(), NULL, JackPortIsInput);
- if ( ports == NULL) {
- errorText_ = "RtApiJack::startStream(): error determining available JACK input ports!";
+ if ( ports[ stream_.channelOffset[0] + i ] )
+ result = jack_connect( handle->client, jack_port_name( handle->ports[0][i] ), ports[ stream_.channelOffset[0] + i ] );
+ if ( result ) {
+ free( ports );
+ errorText_ = "RtApiJack::startStream(): error connecting output ports!";
goto unlock;
}
+ }
+ free(ports);
+ }
- // Now make the port connections. Since RtAudio wasn't designed to
- // allow the user to select particular channels of a device, we'll
- // just open the first "nChannels" ports with offset.
- for ( unsigned int i=0; i<stream_.nUserChannels[0]; i++ ) {
- result = 1;
- if ( ports[ stream_.channelOffset[0] + i ] )
- result = jack_connect( handle->client, jack_port_name( handle->ports[0][i] ), ports[ stream_.channelOffset[0] + i ] );
- if ( result ) {
- free( ports );
- errorText_ = "RtApiJack::startStream(): error connecting output ports!";
- goto unlock;
- }
- }
- free(ports);
+ if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {
+ result = 1;
+ ports = jack_get_ports( handle->client, handle->deviceName[1].c_str(), NULL, JackPortIsOutput );
+ if ( ports == NULL) {
+ errorText_ = "RtApiJack::startStream(): error determining available JACK output ports!";
+ goto unlock;
}
- if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {
+ // Now make the port connections. See note above.
+ for ( unsigned int i=0; i<stream_.nUserChannels[1]; i++ ) {
result = 1;
- ports = jack_get_ports( handle->client, handle->deviceName[1].c_str(), NULL, JackPortIsOutput );
- if ( ports == NULL) {
- errorText_ = "RtApiJack::startStream(): error determining available JACK output ports!";
+ if ( ports[ stream_.channelOffset[1] + i ] )
+ result = jack_connect( handle->client, ports[ stream_.channelOffset[1] + i ], jack_port_name( handle->ports[1][i] ) );
+ if ( result ) {
+ free( ports );
+ errorText_ = "RtApiJack::startStream(): error connecting input ports!";
goto unlock;
}
-
- // Now make the port connections. See note above.
- for ( unsigned int i=0; i<stream_.nUserChannels[1]; i++ ) {
- result = 1;
- if ( ports[ stream_.channelOffset[1] + i ] )
- result = jack_connect( handle->client, ports[ stream_.channelOffset[1] + i ], jack_port_name( handle->ports[1][i] ) );
- if ( result ) {
- free( ports );
- errorText_ = "RtApiJack::startStream(): error connecting input ports!";
- goto unlock;
- }
- }
- free(ports);
}
-
- handle->drainCounter = 0;
- handle->internalDrain = false;
- stream_.state = STREAM_RUNNING;
-
- unlock:
- MUTEX_UNLOCK(&stream_.mutex);
-
- if ( result == 0 ) return;
- error( RtError::SYSTEM_ERROR );
+ free(ports);
}
- void RtApiJack :: stopStream( void )
- {
- verifyStream();
- if ( stream_.state == STREAM_STOPPED ) {
- errorText_ = "RtApiJack::stopStream(): the stream is already stopped!";
- error( RtError::WARNING );
- return;
- }
-
- MUTEX_LOCK( &stream_.mutex );
-
- JackHandle *handle = (JackHandle *) stream_.apiHandle;
- if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
+ handle->drainCounter = 0;
+ handle->internalDrain = false;
+ stream_.state = STREAM_RUNNING;
- if ( handle->drainCounter == 0 ) {
- handle->drainCounter = 1;
- pthread_cond_wait( &handle->condition, &stream_.mutex ); // block until signaled
- }
- }
+ unlock:
+ MUTEX_UNLOCK(&stream_.mutex);
- jack_deactivate( handle->client );
- stream_.state = STREAM_STOPPED;
+ if ( result == 0 ) return;
+ error( RtError::SYSTEM_ERROR );
+}
- MUTEX_UNLOCK( &stream_.mutex );
+void RtApiJack :: stopStream( void )
+{
+ verifyStream();
+ if ( stream_.state == STREAM_STOPPED ) {
+ errorText_ = "RtApiJack::stopStream(): the stream is already stopped!";
+ error( RtError::WARNING );
+ return;
}
- void RtApiJack :: abortStream( void )
- {
- verifyStream();
- if ( stream_.state == STREAM_STOPPED ) {
- errorText_ = "RtApiJack::abortStream(): the stream is already stopped!";
- error( RtError::WARNING );
- return;
- }
-
- JackHandle *handle = (JackHandle *) stream_.apiHandle;
- handle->drainCounter = 1;
+ MUTEX_LOCK( &stream_.mutex );
- stopStream();
+ if ( stream_.state == STREAM_STOPPED ) {
+ MUTEX_UNLOCK( &stream_.mutex );
+ return;
}
- bool RtApiJack :: callbackEvent( unsigned long nframes )
- {
- if ( stream_.state == STREAM_STOPPED ) return SUCCESS;
- if ( stream_.state == STREAM_CLOSED ) {
- errorText_ = "RtApiCore::callbackEvent(): the stream is closed ... this shouldn't happen!";
- error( RtError::WARNING );
- return FAILURE;
- }
- if ( stream_.bufferSize != nframes ) {
- errorText_ = "RtApiCore::callbackEvent(): the JACK buffer size has changed ... cannot process!";
- error( RtError::WARNING );
- return FAILURE;
- }
-
- CallbackInfo *info = (CallbackInfo *) &stream_.callbackInfo;
- JackHandle *handle = (JackHandle *) stream_.apiHandle;
+ JackHandle *handle = (JackHandle *) stream_.apiHandle;
+ if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
- // Check if we were draining the stream and signal is finished.
- if ( handle->drainCounter > 3 ) {
- if ( handle->internalDrain == false )
- pthread_cond_signal( &handle->condition );
- else
- stopStream();
- return SUCCESS;
+ if ( handle->drainCounter == 0 ) {
+ handle->drainCounter = 1;
+ pthread_cond_wait( &handle->condition, &stream_.mutex ); // block until signaled
}
+ }
- MUTEX_LOCK( &stream_.mutex );
+ jack_deactivate( handle->client );
+ stream_.state = STREAM_STOPPED;
- // Invoke user callback first, to get fresh output data.
- if ( handle->drainCounter == 0 ) {
- RtAudioCallback callback = (RtAudioCallback) info->callback;
- double streamTime = getStreamTime();
- RtAudioStreamStatus status = 0;
- if ( stream_.mode != INPUT && handle->xrun[0] == true ) {
- status |= RTAUDIO_OUTPUT_UNDERFLOW;
- handle->xrun[0] = false;
- }
- if ( stream_.mode != OUTPUT && handle->xrun[1] == true ) {
- status |= RTAUDIO_INPUT_OVERFLOW;
- handle->xrun[1] = false;
- }
- handle->drainCounter = callback( stream_.userBuffer[0], stream_.userBuffer[1],
- stream_.bufferSize, streamTime, status, info->userData );
- if ( handle->drainCounter == 2 ) {
- MUTEX_UNLOCK( &stream_.mutex );
- abortStream();
- return SUCCESS;
- }
- else if ( handle->drainCounter == 1 )
- handle->internalDrain = true;
- }
+ MUTEX_UNLOCK( &stream_.mutex );
+}
- jack_default_audio_sample_t *jackbuffer;
- unsigned long bufferBytes = nframes * sizeof( jack_default_audio_sample_t );
- if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
+void RtApiJack :: abortStream( void )
+{
+ verifyStream();
+ if ( stream_.state == STREAM_STOPPED ) {
+ errorText_ = "RtApiJack::abortStream(): the stream is already stopped!";
+ error( RtError::WARNING );
+ return;
+ }
- if ( handle->drainCounter > 0 ) { // write zeros to the output stream
+ JackHandle *handle = (JackHandle *) stream_.apiHandle;
+ handle->drainCounter = 1;
- for ( unsigned int i=0; i<stream_.nDeviceChannels[0]; i++ ) {
- jackbuffer = (jack_default_audio_sample_t *) jack_port_get_buffer( handle->ports[0][i], (jack_nframes_t) nframes );
- memset( jackbuffer, 0, bufferBytes );
- }
+ stopStream();
+}
- }
- else if ( stream_.doConvertBuffer[0] ) {
+// This function will be called by a spawned thread when the user
+// callback function signals that the stream should be stopped or
+// aborted. It is necessary to handle it this way because the
+// callbackEvent() function must return before the jack_deactivate()
+// function will return.
+extern "C" void *jackStopStream( void *ptr )
+{
+ CallbackInfo *info = (CallbackInfo *) ptr;
+ RtApiJack *object = (RtApiJack *) info->object;
- convertBuffer( stream_.deviceBuffer, stream_.userBuffer[0], stream_.convertInfo[0] );
+ object->stopStream();
- for ( unsigned int i=0; i<stream_.nDeviceChannels[0]; i++ ) {
- jackbuffer = (jack_default_audio_sample_t *) jack_port_get_buffer( handle->ports[0][i], (jack_nframes_t) nframes );
- memcpy( jackbuffer, &stream_.deviceBuffer[i*bufferBytes], bufferBytes );
- }
- }
- else { // no buffer conversion
- for ( unsigned int i=0; i<stream_.nUserChannels[0]; i++ ) {
- jackbuffer = (jack_default_audio_sample_t *) jack_port_get_buffer( handle->ports[0][i], (jack_nframes_t) nframes );
- memcpy( jackbuffer, &stream_.userBuffer[0][i*bufferBytes], bufferBytes );
- }
- }
+ pthread_exit( NULL );
+}
- if ( handle->drainCounter ) {
- handle->drainCounter++;
- goto unlock;
- }
- }
+bool RtApiJack :: callbackEvent( unsigned long nframes )
+{
+ if ( stream_.state == STREAM_STOPPED ) return SUCCESS;
+ if ( stream_.state == STREAM_CLOSED ) {
+ errorText_ = "RtApiCore::callbackEvent(): the stream is closed ... this shouldn't happen!";
+ error( RtError::WARNING );
+ return FAILURE;
+ }
+ if ( stream_.bufferSize != nframes ) {
+ errorText_ = "RtApiCore::callbackEvent(): the JACK buffer size has changed ... cannot process!";
+ error( RtError::WARNING );
+ return FAILURE;
+ }
- if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {
+ CallbackInfo *info = (CallbackInfo *) &stream_.callbackInfo;
+ JackHandle *handle = (JackHandle *) stream_.apiHandle;
- if ( stream_.doConvertBuffer[1] ) {
- for ( unsigned int i=0; i<stream_.nDeviceChannels[1]; i++ ) {
- jackbuffer = (jack_default_audio_sample_t *) jack_port_get_buffer( handle->ports[1][i], (jack_nframes_t) nframes );
- memcpy( &stream_.deviceBuffer[i*bufferBytes], jackbuffer, bufferBytes );
- }
- convertBuffer( stream_.userBuffer[1], stream_.deviceBuffer, stream_.convertInfo[1] );
- }
- else { // no buffer conversion
- for ( unsigned int i=0; i<stream_.nUserChannels[1]; i++ ) {
- jackbuffer = (jack_default_audio_sample_t *) jack_port_get_buffer( handle->ports[1][i], (jack_nframes_t) nframes );
- memcpy( &stream_.userBuffer[1][i*bufferBytes], jackbuffer, bufferBytes );
- }
- }
+ // Check if we were draining the stream and signal is finished.
+ if ( handle->drainCounter > 3 ) {
+ if ( handle->internalDrain == true ) {
+ pthread_create( &threadId, NULL, jackStopStream, info );
}
+ else
+ pthread_cond_signal( &handle->condition );
+ return SUCCESS;
+ }
- unlock:
- MUTEX_UNLOCK(&stream_.mutex);
+ MUTEX_LOCK( &stream_.mutex );
- RtApi::tickStreamTime();
+ // The state might change while waiting on a mutex.
+ if ( stream_.state == STREAM_STOPPED ) {
+ MUTEX_UNLOCK( &stream_.mutex );
return SUCCESS;
}
- //******************** End of __UNIX_JACK__ *********************//
-#endif
-#if defined(__WINDOWS_ASIO__) // ASIO API on Windows
+ // Invoke user callback first, to get fresh output data.
+ if ( handle->drainCounter == 0 ) {
+ RtAudioCallback callback = (RtAudioCallback) info->callback;
+ double streamTime = getStreamTime();
+ RtAudioStreamStatus status = 0;
+ if ( stream_.mode != INPUT && handle->xrun[0] == true ) {
+ status |= RTAUDIO_OUTPUT_UNDERFLOW;
+ handle->xrun[0] = false;
+ }
+ if ( stream_.mode != OUTPUT && handle->xrun[1] == true ) {
+ status |= RTAUDIO_INPUT_OVERFLOW;
+ handle->xrun[1] = false;
+ }
+ handle->drainCounter = callback( stream_.userBuffer[0], stream_.userBuffer[1],
+ stream_.bufferSize, streamTime, status, info->userData );
+ if ( handle->drainCounter == 2 ) {
+ MUTEX_UNLOCK( &stream_.mutex );
+ ThreadHandle id;
+ pthread_create( &id, NULL, jackStopStream, info );
+ return SUCCESS;
+ }
+ else if ( handle->drainCounter == 1 )
+ handle->internalDrain = true;
+ }
- // The ASIO API is designed around a callback scheme, so this
- // implementation is similar to that used for OS-X CoreAudio and Linux
- // Jack. The primary constraint with ASIO is that it only allows
- // access to a single driver at a time. Thus, it is not possible to
- // have more than one simultaneous RtAudio stream.
- //
- // This implementation also requires a number of external ASIO files
- // and a few global variables. The ASIO callback scheme does not
- // allow for the passing of user data, so we must create a global
- // pointer to our callbackInfo structure.
- //
- // On unix systems, we make use of a pthread condition variable.
- // Since there is no equivalent in Windows, I hacked something based
- // on information found in
- // http://www.cs.wustl.edu/~schmidt/win32-cv-1.html.
+ jack_default_audio_sample_t *jackbuffer;
+ unsigned long bufferBytes = nframes * sizeof( jack_default_audio_sample_t );
+ if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
-#include "asiosys.h"
-#include "asio.h"
-#include "iasiothiscallresolver.h"
-#include "asiodrivers.h"
-#include <cmath>
+ if ( handle->drainCounter > 0 ) { // write zeros to the output stream
- AsioDrivers drivers;
- ASIOCallbacks asioCallbacks;
- ASIODriverInfo driverInfo;
- CallbackInfo *asioCallbackInfo;
- bool asioXRun;
+ for ( unsigned int i=0; i<stream_.nDeviceChannels[0]; i++ ) {
+ jackbuffer = (jack_default_audio_sample_t *) jack_port_get_buffer( handle->ports[0][i], (jack_nframes_t) nframes );
+ memset( jackbuffer, 0, bufferBytes );
+ }
- struct AsioHandle {
- int drainCounter; // Tracks callback counts when draining
- bool internalDrain; // Indicates if stop is initiated from callback or not.
- ASIOBufferInfo *bufferInfos;
- HANDLE condition;
+ }
+ else if ( stream_.doConvertBuffer[0] ) {
- AsioHandle()
- :drainCounter(0), internalDrain(false), bufferInfos(0) {}
- };
+ convertBuffer( stream_.deviceBuffer, stream_.userBuffer[0], stream_.convertInfo[0] );
- // Function declarations (definitions at end of section)
- static const char* getAsioErrorString( ASIOError result );
- void sampleRateChanged( ASIOSampleRate sRate );
- long asioMessages( long selector, long value, void* message, double* opt );
+ for ( unsigned int i=0; i<stream_.nDeviceChannels[0]; i++ ) {
+ jackbuffer = (jack_default_audio_sample_t *) jack_port_get_buffer( handle->ports[0][i], (jack_nframes_t) nframes );
+ memcpy( jackbuffer, &stream_.deviceBuffer[i*bufferBytes], bufferBytes );
+ }
+ }
+ else { // no buffer conversion
+ for ( unsigned int i=0; i<stream_.nUserChannels[0]; i++ ) {
+ jackbuffer = (jack_default_audio_sample_t *) jack_port_get_buffer( handle->ports[0][i], (jack_nframes_t) nframes );
+ memcpy( jackbuffer, &stream_.userBuffer[0][i*bufferBytes], bufferBytes );
+ }
+ }
- RtApiAsio :: RtApiAsio()
- {
- // ASIO cannot run on a multi-threaded appartment. You can call
- // CoInitialize beforehand, but it must be for appartment threading
- // (in which case, CoInitilialize will return S_FALSE here).
- coInitialized_ = false;
- HRESULT hr = CoInitialize( NULL );
- if ( FAILED(hr) ) {
- errorText_ = "RtApiAsio::ASIO requires a single-threaded appartment. Call CoInitializeEx(0,COINIT_APARTMENTTHREADED)";
- error( RtError::WARNING );
+ if ( handle->drainCounter ) {
+ handle->drainCounter++;
+ goto unlock;
}
- coInitialized_ = true;
+ }
- drivers.removeCurrentDriver();
- driverInfo.asioVersion = 2;
+ if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {
- // See note in DirectSound implementation about GetDesktopWindow().
- driverInfo.sysRef = GetForegroundWindow();
+ if ( stream_.doConvertBuffer[1] ) {
+ for ( unsigned int i=0; i<stream_.nDeviceChannels[1]; i++ ) {
+ jackbuffer = (jack_default_audio_sample_t *) jack_port_get_buffer( handle->ports[1][i], (jack_nframes_t) nframes );
+ memcpy( &stream_.deviceBuffer[i*bufferBytes], jackbuffer, bufferBytes );
+ }
+ convertBuffer( stream_.userBuffer[1], stream_.deviceBuffer, stream_.convertInfo[1] );
+ }
+ else { // no buffer conversion
+ for ( unsigned int i=0; i<stream_.nUserChannels[1]; i++ ) {
+ jackbuffer = (jack_default_audio_sample_t *) jack_port_get_buffer( handle->ports[1][i], (jack_nframes_t) nframes );
+ memcpy( &stream_.userBuffer[1][i*bufferBytes], jackbuffer, bufferBytes );
+ }
+ }
}
- RtApiAsio :: ~RtApiAsio()
- {
- if ( stream_.state != STREAM_CLOSED ) closeStream();
- if ( coInitialized_ ) CoUninitialize();
- }
+ unlock:
+ MUTEX_UNLOCK(&stream_.mutex);
- unsigned int RtApiAsio :: getDeviceCount( void )
- {
- return (unsigned int) drivers.asioGetNumDev();
- }
+ RtApi::tickStreamTime();
+ return SUCCESS;
+}
+ //******************** End of __UNIX_JACK__ *********************//
+#endif
- RtAudio::DeviceInfo RtApiAsio :: getDeviceInfo( unsigned int device )
- {
- RtAudio::DeviceInfo info;
- info.probed = false;
+#if defined(__WINDOWS_ASIO__) // ASIO API on Windows
- // Get device ID
- unsigned int nDevices = getDeviceCount();
- if ( nDevices == 0 ) {
- errorText_ = "RtApiAsio::getDeviceInfo: no devices found!";
- error( RtError::INVALID_USE );
- }
+// The ASIO API is designed around a callback scheme, so this
+// implementation is similar to that used for OS-X CoreAudio and Linux
+// Jack. The primary constraint with ASIO is that it only allows
+// access to a single driver at a time. Thus, it is not possible to
+// have more than one simultaneous RtAudio stream.
+//
+// This implementation also requires a number of external ASIO files
+// and a few global variables. The ASIO callback scheme does not
+// allow for the passing of user data, so we must create a global
+// pointer to our callbackInfo structure.
+//
+// On unix systems, we make use of a pthread condition variable.
+// Since there is no equivalent in Windows, I hacked something based
+// on information found in
+// http://www.cs.wustl.edu/~schmidt/win32-cv-1.html.
- if ( device >= nDevices ) {
- errorText_ = "RtApiAsio::getDeviceInfo: device ID is invalid!";
- error( RtError::INVALID_USE );
- }
+#include "asiosys.h"
+#include "asio.h"
+#include "iasiothiscallresolver.h"
+#include "asiodrivers.h"
+#include <cmath>
- // If a stream is already open, we cannot probe other devices. Thus, use the saved results.
- if ( stream_.state != STREAM_CLOSED ) {
- if ( device >= devices_.size() ) {
- errorText_ = "RtApiAsio::getDeviceInfo: device ID was not present before stream was opened.";
- error( RtError::WARNING );
- return info;
- }
- return devices_[ device ];
- }
+AsioDrivers drivers;
+ASIOCallbacks asioCallbacks;
+ASIODriverInfo driverInfo;
+CallbackInfo *asioCallbackInfo;
+bool asioXRun;
- char driverName[32];
- ASIOError result = drivers.asioGetDriverName( (int) device, driverName, 32 );
- if ( result != ASE_OK ) {
- errorStream_ << "RtApiAsio::getDeviceInfo: unable to get driver name (" << getAsioErrorString( result ) << ").";
- errorText_ = errorStream_.str();
- error( RtError::WARNING );
- return info;
- }
+struct AsioHandle {
+ int drainCounter; // Tracks callback counts when draining
+ bool internalDrain; // Indicates if stop is initiated from callback or not.
+ ASIOBufferInfo *bufferInfos;
+ HANDLE condition;
- info.name = driverName;
+ AsioHandle()
+ :drainCounter(0), internalDrain(false), bufferInfos(0) {}
+};
- if ( !drivers.loadDriver( driverName ) ) {
- errorStream_ << "RtApiAsio::getDeviceInfo: unable to load driver (" << driverName << ").";
- errorText_ = errorStream_.str();
- error( RtError::WARNING );
- return info;
- }
+// Function declarations (definitions at end of section)
+static const char* getAsioErrorString( ASIOError result );
+void sampleRateChanged( ASIOSampleRate sRate );
+long asioMessages( long selector, long value, void* message, double* opt );
- result = ASIOInit( &driverInfo );
- if ( result != ASE_OK ) {
- errorStream_ << "RtApiAsio::getDeviceInfo: error (" << getAsioErrorString( result ) << ") initializing driver (" << driverName << ").";
- errorText_ = errorStream_.str();
- error( RtError::WARNING );
- return info;
- }
+RtApiAsio :: RtApiAsio()
+{
+ // ASIO cannot run on a multi-threaded appartment. You can call
+ // CoInitialize beforehand, but it must be for appartment threading
+ // (in which case, CoInitilialize will return S_FALSE here).
+ coInitialized_ = false;
+ HRESULT hr = CoInitialize( NULL );
+ if ( FAILED(hr) ) {
+ errorText_ = "RtApiAsio::ASIO requires a single-threaded appartment. Call CoInitializeEx(0,COINIT_APARTMENTTHREADED)";
+ error( RtError::WARNING );
+ }
+ coInitialized_ = true;
- // Determine the device channel information.
- long inputChannels, outputChannels;
- result = ASIOGetChannels( &inputChannels, &outputChannels );
- if ( result != ASE_OK ) {
- drivers.removeCurrentDriver();
- errorStream_ << "RtApiAsio::getDeviceInfo: error (" << getAsioErrorString( result ) << ") getting channel count (" << driverName << ").";
- errorText_ = errorStream_.str();
- error( RtError::WARNING );
- return info;
- }
+ drivers.removeCurrentDriver();
+ driverInfo.asioVersion = 2;
- info.outputChannels = outputChannels;
- info.inputChannels = inputChannels;
- if ( info.outputChannels > 0 && info.inputChannels > 0 )
- info.duplexChannels = (info.outputChannels > info.inputChannels) ? info.inputChannels : info.outputChannels;
+ // See note in DirectSound implementation about GetDesktopWindow().
+ driverInfo.sysRef = GetForegroundWindow();
+}
- // Determine the supported sample rates.
- info.sampleRates.clear();
- for ( unsigned int i=0; i<MAX_SAMPLE_RATES; i++ ) {
- result = ASIOCanSampleRate( (ASIOSampleRate) SAMPLE_RATES[i] );
- if ( result == ASE_OK )
- info.sampleRates.push_back( SAMPLE_RATES[i] );
- }
+RtApiAsio :: ~RtApiAsio()
+{
+ if ( stream_.state != STREAM_CLOSED ) closeStream();
+ if ( coInitialized_ ) CoUninitialize();
+}
- // Determine supported data types ... just check first channel and assume rest are the same.
- ASIOChannelInfo channelInfo;
- channelInfo.channel = 0;
- channelInfo.isInput = true;
- if ( info.inputChannels <= 0 ) channelInfo.isInput = false;
- result = ASIOGetChannelInfo( &channelInfo );
- if ( result != ASE_OK ) {
- drivers.removeCurrentDriver();
- errorStream_ << "RtApiAsio::getDeviceInfo: error (" << getAsioErrorString( result ) << ") getting driver channel info (" << driverName << ").";
- errorText_ = errorStream_.str();
+unsigned int RtApiAsio :: getDeviceCount( void )
+{
+ return (unsigned int) drivers.asioGetNumDev();
+}
+
+RtAudio::DeviceInfo RtApiAsio :: getDeviceInfo( unsigned int device )
+{
+ RtAudio::DeviceInfo info;
+ info.probed = false;
+
+ // Get device ID
+ unsigned int nDevices = getDeviceCount();
+ if ( nDevices == 0 ) {
+ errorText_ = "RtApiAsio::getDeviceInfo: no devices found!";
+ error( RtError::INVALID_USE );
+ }
+
+ if ( device >= nDevices ) {
+ errorText_ = "RtApiAsio::getDeviceInfo: device ID is invalid!";
+ error( RtError::INVALID_USE );
+ }
+
+ // If a stream is already open, we cannot probe other devices. Thus, use the saved results.
+ if ( stream_.state != STREAM_CLOSED ) {
+ if ( device >= devices_.size() ) {
+ errorText_ = "RtApiAsio::getDeviceInfo: device ID was not present before stream was opened.";
error( RtError::WARNING );
return info;
}
+ return devices_[ device ];
+ }
- info.nativeFormats = 0;
- if ( channelInfo.type == ASIOSTInt16MSB || channelInfo.type == ASIOSTInt16LSB )
- info.nativeFormats |= RTAUDIO_SINT16;
- else if ( channelInfo.type == ASIOSTInt32MSB || channelInfo.type == ASIOSTInt32LSB )
- info.nativeFormats |= RTAUDIO_SINT32;
- else if ( channelInfo.type == ASIOSTFloat32MSB || channelInfo.type == ASIOSTFloat32LSB )
- info.nativeFormats |= RTAUDIO_FLOAT32;
- else if ( channelInfo.type == ASIOSTFloat64MSB || channelInfo.type == ASIOSTFloat64LSB )
- info.nativeFormats |= RTAUDIO_FLOAT64;
+ char driverName[32];
+ ASIOError result = drivers.asioGetDriverName( (int) device, driverName, 32 );
+ if ( result != ASE_OK ) {
+ errorStream_ << "RtApiAsio::getDeviceInfo: unable to get driver name (" << getAsioErrorString( result ) << ").";
+ errorText_ = errorStream_.str();
+ error( RtError::WARNING );
+ return info;
+ }
- if ( getDefaultOutputDevice() == device )
- info.isDefaultOutput = true;
- if ( getDefaultInputDevice() == device )
- info.isDefaultInput = true;
+ info.name = driverName;
- info.probed = true;
- drivers.removeCurrentDriver();
+ if ( !drivers.loadDriver( driverName ) ) {
+ errorStream_ << "RtApiAsio::getDeviceInfo: unable to load driver (" << driverName << ").";
+ errorText_ = errorStream_.str();
+ error( RtError::WARNING );
return info;
}
- void bufferSwitch( long index, ASIOBool processNow )
- {
- RtApiAsio *object = (RtApiAsio *) asioCallbackInfo->object;
- object->callbackEvent( index );
+ result = ASIOInit( &driverInfo );
+ if ( result != ASE_OK ) {
+ errorStream_ << "RtApiAsio::getDeviceInfo: error (" << getAsioErrorString( result ) << ") initializing driver (" << driverName << ").";
+ errorText_ = errorStream_.str();
+ error( RtError::WARNING );
+ return info;
}
- void RtApiAsio :: saveDeviceInfo( void )
- {
- devices_.clear();
+ // Determine the device channel information.
+ long inputChannels, outputChannels;
+ result = ASIOGetChannels( &inputChannels, &outputChannels );
+ if ( result != ASE_OK ) {
+ drivers.removeCurrentDriver();
+ errorStream_ << "RtApiAsio::getDeviceInfo: error (" << getAsioErrorString( result ) << ") getting channel count (" << driverName << ").";
+ errorText_ = errorStream_.str();
+ error( RtError::WARNING );
+ return info;
+ }
+
+ info.outputChannels = outputChannels;
+ info.inputChannels = inputChannels;
+ if ( info.outputChannels > 0 && info.inputChannels > 0 )
+ info.duplexChannels = (info.outputChannels > info.inputChannels) ? info.inputChannels : info.outputChannels;
- unsigned int nDevices = getDeviceCount();
- devices_.resize( nDevices );
- for ( unsigned int i=0; i<nDevices; i++ )
- devices_[i] = getDeviceInfo( i );
+ // Determine the supported sample rates.
+ info.sampleRates.clear();
+ for ( unsigned int i=0; i<MAX_SAMPLE_RATES; i++ ) {
+ result = ASIOCanSampleRate( (ASIOSampleRate) SAMPLE_RATES[i] );
+ if ( result == ASE_OK )
+ info.sampleRates.push_back( SAMPLE_RATES[i] );
+ }
+
+ // Determine supported data types ... just check first channel and assume rest are the same.
+ ASIOChannelInfo channelInfo;
+ channelInfo.channel = 0;
+ channelInfo.isInput = true;
+ if ( info.inputChannels <= 0 ) channelInfo.isInput = false;
+ result = ASIOGetChannelInfo( &channelInfo );
+ if ( result != ASE_OK ) {
+ drivers.removeCurrentDriver();
+ errorStream_ << "RtApiAsio::getDeviceInfo: error (" << getAsioErrorString( result ) << ") getting driver channel info (" << driverName << ").";
+ errorText_ = errorStream_.str();
+ error( RtError::WARNING );
+ return info;
}
- bool RtApiAsio :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,
- unsigned int firstChannel, unsigned int sampleRate,
- RtAudioFormat format, unsigned int *bufferSize,
- RtAudio::StreamOptions *options )
- {
- // For ASIO, a duplex stream MUST use the same driver.
- if ( mode == INPUT && stream_.mode == OUTPUT && stream_.device[0] != device ) {
- errorText_ = "RtApiAsio::probeDeviceOpen: an ASIO duplex stream must use the same device for input and output!";
- return FAILURE;
- }
+ info.nativeFormats = 0;
+ if ( channelInfo.type == ASIOSTInt16MSB || channelInfo.type == ASIOSTInt16LSB )
+ info.nativeFormats |= RTAUDIO_SINT16;
+ else if ( channelInfo.type == ASIOSTInt32MSB || channelInfo.type == ASIOSTInt32LSB )
+ info.nativeFormats |= RTAUDIO_SINT32;
+ else if ( channelInfo.type == ASIOSTFloat32MSB || channelInfo.type == ASIOSTFloat32LSB )
+ info.nativeFormats |= RTAUDIO_FLOAT32;
+ else if ( channelInfo.type == ASIOSTFloat64MSB || channelInfo.type == ASIOSTFloat64LSB )
+ info.nativeFormats |= RTAUDIO_FLOAT64;
- char driverName[32];
- ASIOError result = drivers.asioGetDriverName( (int) device, driverName, 32 );
- if ( result != ASE_OK ) {
- errorStream_ << "RtApiAsio::probeDeviceOpen: unable to get driver name (" << getAsioErrorString( result ) << ").";
- errorText_ = errorStream_.str();
- return FAILURE;
- }
+ if ( info.outputChannels > 0 )
+ if ( getDefaultOutputDevice() == device ) info.isDefaultOutput = true;
+ if ( info.inputChannels > 0 )
+ if ( getDefaultInputDevice() == device ) info.isDefaultInput = true;
- // The getDeviceInfo() function will not work when a stream is open
- // because ASIO does not allow multiple devices to run at the same
- // time. Thus, we'll probe the system before opening a stream and
- // save the results for use by getDeviceInfo().
- this->saveDeviceInfo();
+ info.probed = true;
+ drivers.removeCurrentDriver();
+ return info;
+}
- // Only load the driver once for duplex stream.
- if ( mode != INPUT || stream_.mode != OUTPUT ) {
- if ( !drivers.loadDriver( driverName ) ) {
- errorStream_ << "RtApiAsio::probeDeviceOpen: unable to load driver (" << driverName << ").";
- errorText_ = errorStream_.str();
- return FAILURE;
- }
+void bufferSwitch( long index, ASIOBool processNow )
+{
+ RtApiAsio *object = (RtApiAsio *) asioCallbackInfo->object;
+ object->callbackEvent( index );
+}
- result = ASIOInit( &driverInfo );
- if ( result != ASE_OK ) {
- errorStream_ << "RtApiAsio::probeDeviceOpen: error (" << getAsioErrorString( result ) << ") initializing driver (" << driverName << ").";
- errorText_ = errorStream_.str();
- return FAILURE;
- }
- }
+void RtApiAsio :: saveDeviceInfo( void )
+{
+ devices_.clear();
- // Check the device channel count.
- long inputChannels, outputChannels;
- result = ASIOGetChannels( &inputChannels, &outputChannels );
- if ( result != ASE_OK ) {
- drivers.removeCurrentDriver();
- errorStream_ << "RtApiAsio::probeDeviceOpen: error (" << getAsioErrorString( result ) << ") getting channel count (" << driverName << ").";
- errorText_ = errorStream_.str();
- return FAILURE;
- }
+ unsigned int nDevices = getDeviceCount();
+ devices_.resize( nDevices );
+ for ( unsigned int i=0; i<nDevices; i++ )
+ devices_[i] = getDeviceInfo( i );
+}
- if ( ( mode == OUTPUT && (channels+firstChannel) > (unsigned int) outputChannels) ||
- ( mode == INPUT && (channels+firstChannel) > (unsigned int) inputChannels) ) {
- drivers.removeCurrentDriver();
- errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") does not support requested channel count (" << channels << ") + offset (" << firstChannel << ").";
- errorText_ = errorStream_.str();
- return FAILURE;
- }
- stream_.nDeviceChannels[mode] = channels;
- stream_.nUserChannels[mode] = channels;
- stream_.channelOffset[mode] = firstChannel;
+bool RtApiAsio :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,
+ unsigned int firstChannel, unsigned int sampleRate,
+ RtAudioFormat format, unsigned int *bufferSize,
+ RtAudio::StreamOptions *options )
+{
+ // For ASIO, a duplex stream MUST use the same driver.
+ if ( mode == INPUT && stream_.mode == OUTPUT && stream_.device[0] != device ) {
+ errorText_ = "RtApiAsio::probeDeviceOpen: an ASIO duplex stream must use the same device for input and output!";
+ return FAILURE;
+ }
- // Verify the sample rate is supported.
- result = ASIOCanSampleRate( (ASIOSampleRate) sampleRate );
- if ( result != ASE_OK ) {
- drivers.removeCurrentDriver();
- errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") does not support requested sample rate (" << sampleRate << ").";
+ char driverName[32];
+ ASIOError result = drivers.asioGetDriverName( (int) device, driverName, 32 );
+ if ( result != ASE_OK ) {
+ errorStream_ << "RtApiAsio::probeDeviceOpen: unable to get driver name (" << getAsioErrorString( result ) << ").";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
+
+ // The getDeviceInfo() function will not work when a stream is open
+ // because ASIO does not allow multiple devices to run at the same
+ // time. Thus, we'll probe the system before opening a stream and
+ // save the results for use by getDeviceInfo().
+ this->saveDeviceInfo();
+
+ // Only load the driver once for duplex stream.
+ if ( mode != INPUT || stream_.mode != OUTPUT ) {
+ if ( !drivers.loadDriver( driverName ) ) {
+ errorStream_ << "RtApiAsio::probeDeviceOpen: unable to load driver (" << driverName << ").";
errorText_ = errorStream_.str();
return FAILURE;
}
- // Get the current sample rate
- ASIOSampleRate currentRate;
- result = ASIOGetSampleRate( ¤tRate );
+ result = ASIOInit( &driverInfo );
if ( result != ASE_OK ) {
- drivers.removeCurrentDriver();
- errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error getting sample rate.";
+ errorStream_ << "RtApiAsio::probeDeviceOpen: error (" << getAsioErrorString( result ) << ") initializing driver (" << driverName << ").";
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;
- }
- }
+ // Check the device channel count.
+ long inputChannels, outputChannels;
+ result = ASIOGetChannels( &inputChannels, &outputChannels );
+ if ( result != ASE_OK ) {
+ drivers.removeCurrentDriver();
+ errorStream_ << "RtApiAsio::probeDeviceOpen: error (" << getAsioErrorString( result ) << ") getting channel count (" << driverName << ").";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
- // Determine the driver data type.
- ASIOChannelInfo channelInfo;
- channelInfo.channel = 0;
- if ( mode == OUTPUT ) channelInfo.isInput = false;
- else channelInfo.isInput = true;
- result = ASIOGetChannelInfo( &channelInfo );
- if ( result != ASE_OK ) {
- drivers.removeCurrentDriver();
- errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error (" << getAsioErrorString( result ) << ") getting data format.";
- errorText_ = errorStream_.str();
- return FAILURE;
- }
+ if ( ( mode == OUTPUT && (channels+firstChannel) > (unsigned int) outputChannels) ||
+ ( mode == INPUT && (channels+firstChannel) > (unsigned int) inputChannels) ) {
+ drivers.removeCurrentDriver();
+ errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") does not support requested channel count (" << channels << ") + offset (" << firstChannel << ").";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
+ stream_.nDeviceChannels[mode] = channels;
+ stream_.nUserChannels[mode] = channels;
+ stream_.channelOffset[mode] = firstChannel;
- // Assuming WINDOWS host is always little-endian.
- stream_.doByteSwap[mode] = false;
- stream_.userFormat = format;
- stream_.deviceFormat[mode] = 0;
- if ( channelInfo.type == ASIOSTInt16MSB || channelInfo.type == ASIOSTInt16LSB ) {
- stream_.deviceFormat[mode] = RTAUDIO_SINT16;
- if ( channelInfo.type == ASIOSTInt16MSB ) stream_.doByteSwap[mode] = true;
- }
- else if ( channelInfo.type == ASIOSTInt32MSB || channelInfo.type == ASIOSTInt32LSB ) {
- stream_.deviceFormat[mode] = RTAUDIO_SINT32;
- if ( channelInfo.type == ASIOSTInt32MSB ) stream_.doByteSwap[mode] = true;
- }
- else if ( channelInfo.type == ASIOSTFloat32MSB || channelInfo.type == ASIOSTFloat32LSB ) {
- stream_.deviceFormat[mode] = RTAUDIO_FLOAT32;
- if ( channelInfo.type == ASIOSTFloat32MSB ) stream_.doByteSwap[mode] = true;
- }
- else if ( channelInfo.type == ASIOSTFloat64MSB || channelInfo.type == ASIOSTFloat64LSB ) {
- stream_.deviceFormat[mode] = RTAUDIO_FLOAT64;
- if ( channelInfo.type == ASIOSTFloat64MSB ) stream_.doByteSwap[mode] = true;
- }
+ // Verify the sample rate is supported.
+ result = ASIOCanSampleRate( (ASIOSampleRate) sampleRate );
+ if ( result != ASE_OK ) {
+ drivers.removeCurrentDriver();
+ errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") does not support requested sample rate (" << sampleRate << ").";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
- if ( stream_.deviceFormat[mode] == 0 ) {
- drivers.removeCurrentDriver();
- errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") data format not supported by RtAudio.";
- errorText_ = errorStream_.str();
- return FAILURE;
- }
+ // Get the current sample rate
+ ASIOSampleRate currentRate;
+ result = ASIOGetSampleRate( ¤tRate );
+ if ( result != ASE_OK ) {
+ drivers.removeCurrentDriver();
+ errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error getting sample rate.";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
- // Set the buffer size. For a duplex stream, this will end up
- // setting the buffer size based on the input constraints, which
- // should be ok.
- long minSize, maxSize, preferSize, granularity;
- result = ASIOGetBufferSize( &minSize, &maxSize, &preferSize, &granularity );
+ // 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 (" << getAsioErrorString( result ) << ") getting buffer size.";
+ errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error setting sample rate (" << sampleRate << ").";
errorText_ = errorStream_.str();
return FAILURE;
}
+ }
- if ( *bufferSize < (unsigned int) minSize ) *bufferSize = (unsigned int) minSize;
- else if ( *bufferSize > (unsigned int) maxSize ) *bufferSize = (unsigned int) maxSize;
- else if ( granularity == -1 ) {
- // Make sure bufferSize is a power of two.
- int log2_of_min_size = 0;
- int log2_of_max_size = 0;
-
- for ( unsigned int i = 0; i < sizeof(long) * 8; i++ ) {
- if ( minSize & ((long)1 << i) ) log2_of_min_size = i;
- if ( maxSize & ((long)1 << i) ) log2_of_max_size = i;
- }
-
- long min_delta = std::abs( (long)*bufferSize - ((long)1 << log2_of_min_size) );
- int min_delta_num = log2_of_min_size;
+ // Determine the driver data type.
+ ASIOChannelInfo channelInfo;
+ channelInfo.channel = 0;
+ if ( mode == OUTPUT ) channelInfo.isInput = false;
+ else channelInfo.isInput = true;
+ result = ASIOGetChannelInfo( &channelInfo );
+ if ( result != ASE_OK ) {
+ drivers.removeCurrentDriver();
+ errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error (" << getAsioErrorString( result ) << ") getting data format.";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
- for (int i = log2_of_min_size + 1; i <= log2_of_max_size; i++) {
- long current_delta = std::abs( (long)*bufferSize - ((long)1 << i) );
- if (current_delta < min_delta) {
- min_delta = current_delta;
- min_delta_num = i;
- }
- }
-
- *bufferSize = ( (unsigned int)1 << min_delta_num );
- if ( *bufferSize < (unsigned int) minSize ) *bufferSize = (unsigned int) minSize;
- else if ( *bufferSize > (unsigned int) maxSize ) *bufferSize = (unsigned int) maxSize;
- }
- else if ( granularity != 0 ) {
- // Set to an even multiple of granularity, rounding up.
- *bufferSize = (*bufferSize + granularity-1) / granularity * granularity;
- }
-
- if ( mode == INPUT && stream_.mode == OUTPUT && stream_.bufferSize != *bufferSize ) {
- drivers.removeCurrentDriver();
- errorText_ = "RtApiAsio::probeDeviceOpen: input/output buffersize discrepancy!";
- return FAILURE;
- }
+ // Assuming WINDOWS host is always little-endian.
+ stream_.doByteSwap[mode] = false;
+ stream_.userFormat = format;
+ stream_.deviceFormat[mode] = 0;
+ if ( channelInfo.type == ASIOSTInt16MSB || channelInfo.type == ASIOSTInt16LSB ) {
+ stream_.deviceFormat[mode] = RTAUDIO_SINT16;
+ if ( channelInfo.type == ASIOSTInt16MSB ) stream_.doByteSwap[mode] = true;
+ }
+ else if ( channelInfo.type == ASIOSTInt32MSB || channelInfo.type == ASIOSTInt32LSB ) {
+ stream_.deviceFormat[mode] = RTAUDIO_SINT32;
+ if ( channelInfo.type == ASIOSTInt32MSB ) stream_.doByteSwap[mode] = true;
+ }
+ else if ( channelInfo.type == ASIOSTFloat32MSB || channelInfo.type == ASIOSTFloat32LSB ) {
+ stream_.deviceFormat[mode] = RTAUDIO_FLOAT32;
+ if ( channelInfo.type == ASIOSTFloat32MSB ) stream_.doByteSwap[mode] = true;
+ }
+ else if ( channelInfo.type == ASIOSTFloat64MSB || channelInfo.type == ASIOSTFloat64LSB ) {
+ stream_.deviceFormat[mode] = RTAUDIO_FLOAT64;
+ if ( channelInfo.type == ASIOSTFloat64MSB ) stream_.doByteSwap[mode] = true;
+ }
- stream_.bufferSize = *bufferSize;
- stream_.nBuffers = 2;
+ if ( stream_.deviceFormat[mode] == 0 ) {
+ drivers.removeCurrentDriver();
+ errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") data format not supported by RtAudio.";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
- if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) stream_.userInterleaved = false;
- else stream_.userInterleaved = true;
+ // Set the buffer size. For a duplex stream, this will end up
+ // setting the buffer size based on the input constraints, which
+ // should be ok.
+ long minSize, maxSize, preferSize, granularity;
+ result = ASIOGetBufferSize( &minSize, &maxSize, &preferSize, &granularity );
+ if ( result != ASE_OK ) {
+ drivers.removeCurrentDriver();
+ errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error (" << getAsioErrorString( result ) << ") getting buffer size.";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
- // ASIO always uses non-interleaved buffers.
- stream_.deviceInterleaved[mode] = false;
+ if ( *bufferSize < (unsigned int) minSize ) *bufferSize = (unsigned int) minSize;
+ else if ( *bufferSize > (unsigned int) maxSize ) *bufferSize = (unsigned int) maxSize;
+ else if ( granularity == -1 ) {
+ // Make sure bufferSize is a power of two.
+ int log2_of_min_size = 0;
+ int log2_of_max_size = 0;
- // Allocate, if necessary, our AsioHandle structure for the stream.
- AsioHandle *handle = (AsioHandle *) stream_.apiHandle;
- if ( handle == 0 ) {
- try {
- handle = new AsioHandle;
- }
- catch ( std::bad_alloc& ) {
- //if ( handle == NULL ) {
- drivers.removeCurrentDriver();
- errorText_ = "RtApiAsio::probeDeviceOpen: error allocating AsioHandle memory.";
- return FAILURE;
- }
- handle->bufferInfos = 0;
-
- // Create a manual-reset event.
- handle->condition = CreateEvent( NULL, // no security
- TRUE, // manual-reset
- FALSE, // non-signaled initially
- NULL ); // unnamed
- stream_.apiHandle = (void *) handle;
- }
-
- // Create the ASIO internal buffers. Since RtAudio sets up input
- // and output separately, we'll have to dispose of previously
- // created output buffers for a duplex stream.
- long inputLatency, outputLatency;
- if ( mode == INPUT && stream_.mode == OUTPUT ) {
- ASIODisposeBuffers();
- if ( handle->bufferInfos ) free( handle->bufferInfos );
- }
-
- // Allocate, initialize, and save the bufferInfos in our stream callbackInfo structure.
- bool buffersAllocated = false;
- unsigned int i, nChannels = stream_.nDeviceChannels[0] + stream_.nDeviceChannels[1];
- handle->bufferInfos = (ASIOBufferInfo *) malloc( nChannels * sizeof(ASIOBufferInfo) );
- if ( handle->bufferInfos == NULL ) {
- errorStream_ << "RtApiAsio::probeDeviceOpen: error allocating bufferInfo memory for driver (" << driverName << ").";
- errorText_ = errorStream_.str();
- goto error;
+ for ( unsigned int i = 0; i < sizeof(long) * 8; i++ ) {
+ if ( minSize & ((long)1 << i) ) log2_of_min_size = i;
+ if ( maxSize & ((long)1 << i) ) log2_of_max_size = i;
}
- ASIOBufferInfo *infos;
- infos = handle->bufferInfos;
- for ( i=0; i<stream_.nDeviceChannels[0]; i++, infos++ ) {
- infos->isInput = ASIOFalse;
- infos->channelNum = i + stream_.channelOffset[0];
- infos->buffers[0] = infos->buffers[1] = 0;
- }
- for ( i=0; i<stream_.nDeviceChannels[1]; i++, infos++ ) {
- infos->isInput = ASIOTrue;
- infos->channelNum = i + stream_.channelOffset[1];
- infos->buffers[0] = infos->buffers[1] = 0;
- }
+ long min_delta = std::abs( (long)*bufferSize - ((long)1 << log2_of_min_size) );
+ int min_delta_num = log2_of_min_size;
- // Set up the ASIO callback structure and create the ASIO data buffers.
- asioCallbacks.bufferSwitch = &bufferSwitch;
- asioCallbacks.sampleRateDidChange = &sampleRateChanged;
- asioCallbacks.asioMessage = &asioMessages;
- asioCallbacks.bufferSwitchTimeInfo = NULL;
- result = ASIOCreateBuffers( handle->bufferInfos, nChannels, stream_.bufferSize, &asioCallbacks );
- if ( result != ASE_OK ) {
- errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error (" << getAsioErrorString( result ) << ") creating buffers.";
- errorText_ = errorStream_.str();
- goto error;
+ for (int i = log2_of_min_size + 1; i <= log2_of_max_size; i++) {
+ long current_delta = std::abs( (long)*bufferSize - ((long)1 << i) );
+ if (current_delta < min_delta) {
+ min_delta = current_delta;
+ min_delta_num = i;
+ }
}
- buffersAllocated = true;
-
- // Set flags for buffer conversion.
- stream_.doConvertBuffer[mode] = false;
- if ( stream_.userFormat != stream_.deviceFormat[mode] )
- stream_.doConvertBuffer[mode] = true;
- if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] &&
- stream_.nUserChannels[mode] > 1 )
- stream_.doConvertBuffer[mode] = true;
- // Allocate necessary internal buffers
- 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.";
- goto error;
- }
+ *bufferSize = ( (unsigned int)1 << min_delta_num );
+ if ( *bufferSize < (unsigned int) minSize ) *bufferSize = (unsigned int) minSize;
+ else if ( *bufferSize > (unsigned int) maxSize ) *bufferSize = (unsigned int) maxSize;
+ }
+ else if ( granularity != 0 ) {
+ // Set to an even multiple of granularity, rounding up.
+ *bufferSize = (*bufferSize + granularity-1) / granularity * granularity;
+ }
- if ( stream_.doConvertBuffer[mode] ) {
+ if ( mode == INPUT && stream_.mode == OUTPUT && stream_.bufferSize != *bufferSize ) {
+ drivers.removeCurrentDriver();
+ errorText_ = "RtApiAsio::probeDeviceOpen: input/output buffersize discrepancy!";
+ return FAILURE;
+ }
- bool makeBuffer = true;
- bufferBytes = stream_.nDeviceChannels[mode] * formatBytes( stream_.deviceFormat[mode] );
- if ( mode == INPUT ) {
- if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) {
- unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] );
- if ( bufferBytes <= bytesOut ) makeBuffer = false;
- }
- }
+ stream_.bufferSize = *bufferSize;
+ stream_.nBuffers = 2;
- if ( makeBuffer ) {
- bufferBytes *= *bufferSize;
- if ( stream_.deviceBuffer ) free( stream_.deviceBuffer );
- stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 );
- if ( stream_.deviceBuffer == NULL ) {
- errorText_ = "RtApiAsio::probeDeviceOpen: error allocating device buffer memory.";
- goto error;
- }
- }
- }
+ if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) stream_.userInterleaved = false;
+ else stream_.userInterleaved = true;
- stream_.sampleRate = sampleRate;
- stream_.device[mode] = device;
- stream_.state = STREAM_STOPPED;
- asioCallbackInfo = &stream_.callbackInfo;
- stream_.callbackInfo.object = (void *) this;
- if ( stream_.mode == OUTPUT && mode == INPUT )
- // We had already set up an output stream.
- stream_.mode = DUPLEX;
- else
- stream_.mode = mode;
+ // ASIO always uses non-interleaved buffers.
+ stream_.deviceInterleaved[mode] = false;
- // Determine device latencies
- result = ASIOGetLatencies( &inputLatency, &outputLatency );
- if ( result != ASE_OK ) {
- errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error (" << getAsioErrorString( result ) << ") getting latency.";
- errorText_ = errorStream_.str();
- error( RtError::WARNING); // warn but don't fail
+ // Allocate, if necessary, our AsioHandle structure for the stream.
+ AsioHandle *handle = (AsioHandle *) stream_.apiHandle;
+ if ( handle == 0 ) {
+ try {
+ handle = new AsioHandle;
}
- else {
- stream_.latency[0] = outputLatency;
- stream_.latency[1] = inputLatency;
+ catch ( std::bad_alloc& ) {
+ //if ( handle == NULL ) {
+ drivers.removeCurrentDriver();
+ errorText_ = "RtApiAsio::probeDeviceOpen: error allocating AsioHandle memory.";
+ return FAILURE;
}
+ handle->bufferInfos = 0;
- // Setup the buffer conversion information structure. We don't use
- // buffers to do channel offsets, so we override that parameter
- // here.
- if ( stream_.doConvertBuffer[mode] ) setConvertInfo( mode, 0 );
-
- return SUCCESS;
-
- error:
- if ( buffersAllocated )
- ASIODisposeBuffers();
- drivers.removeCurrentDriver();
+ // Create a manual-reset event.
+ handle->condition = CreateEvent( NULL, // no security
+ TRUE, // manual-reset
+ FALSE, // non-signaled initially
+ NULL ); // unnamed
+ stream_.apiHandle = (void *) handle;
+ }
- if ( handle ) {
- CloseHandle( handle->condition );
- if ( handle->bufferInfos )
- free( handle->bufferInfos );
- delete handle;
- stream_.apiHandle = 0;
- }
+ // Create the ASIO internal buffers. Since RtAudio sets up input
+ // and output separately, we'll have to dispose of previously
+ // created output buffers for a duplex stream.
+ long inputLatency, outputLatency;
+ if ( mode == INPUT && stream_.mode == OUTPUT ) {
+ ASIODisposeBuffers();
+ if ( handle->bufferInfos ) free( handle->bufferInfos );
+ }
- for ( int i=0; i<2; i++ ) {
- if ( stream_.userBuffer[i] ) {
- free( stream_.userBuffer[i] );
- stream_.userBuffer[i] = 0;
- }
- }
+ // Allocate, initialize, and save the bufferInfos in our stream callbackInfo structure.
+ bool buffersAllocated = false;
+ unsigned int i, nChannels = stream_.nDeviceChannels[0] + stream_.nDeviceChannels[1];
+ handle->bufferInfos = (ASIOBufferInfo *) malloc( nChannels * sizeof(ASIOBufferInfo) );
+ if ( handle->bufferInfos == NULL ) {
+ errorStream_ << "RtApiAsio::probeDeviceOpen: error allocating bufferInfo memory for driver (" << driverName << ").";
+ errorText_ = errorStream_.str();
+ goto error;
+ }
- if ( stream_.deviceBuffer ) {
- free( stream_.deviceBuffer );
- stream_.deviceBuffer = 0;
- }
+ ASIOBufferInfo *infos;
+ infos = handle->bufferInfos;
+ for ( i=0; i<stream_.nDeviceChannels[0]; i++, infos++ ) {
+ infos->isInput = ASIOFalse;
+ infos->channelNum = i + stream_.channelOffset[0];
+ infos->buffers[0] = infos->buffers[1] = 0;
+ }
+ for ( i=0; i<stream_.nDeviceChannels[1]; i++, infos++ ) {
+ infos->isInput = ASIOTrue;
+ infos->channelNum = i + stream_.channelOffset[1];
+ infos->buffers[0] = infos->buffers[1] = 0;
+ }
- return FAILURE;
+ // Set up the ASIO callback structure and create the ASIO data buffers.
+ asioCallbacks.bufferSwitch = &bufferSwitch;
+ asioCallbacks.sampleRateDidChange = &sampleRateChanged;
+ asioCallbacks.asioMessage = &asioMessages;
+ asioCallbacks.bufferSwitchTimeInfo = NULL;
+ result = ASIOCreateBuffers( handle->bufferInfos, nChannels, stream_.bufferSize, &asioCallbacks );
+ if ( result != ASE_OK ) {
+ errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error (" << getAsioErrorString( result ) << ") creating buffers.";
+ errorText_ = errorStream_.str();
+ goto error;
}
+ buffersAllocated = true;
- void RtApiAsio :: closeStream()
- {
- if ( stream_.state == STREAM_CLOSED ) {
- errorText_ = "RtApiAsio::closeStream(): no open stream to close!";
- error( RtError::WARNING );
- return;
- }
+ // Set flags for buffer conversion.
+ stream_.doConvertBuffer[mode] = false;
+ if ( stream_.userFormat != stream_.deviceFormat[mode] )
+ stream_.doConvertBuffer[mode] = true;
+ if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] &&
+ stream_.nUserChannels[mode] > 1 )
+ stream_.doConvertBuffer[mode] = true;
- if ( stream_.state == STREAM_RUNNING ) {
- stream_.state = STREAM_STOPPED;
- ASIOStop();
- }
- ASIODisposeBuffers();
- drivers.removeCurrentDriver();
+ // Allocate necessary internal buffers
+ 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.";
+ goto error;
+ }
- AsioHandle *handle = (AsioHandle *) stream_.apiHandle;
- if ( handle ) {
- CloseHandle( handle->condition );
- if ( handle->bufferInfos )
- free( handle->bufferInfos );
- delete handle;
- stream_.apiHandle = 0;
- }
+ if ( stream_.doConvertBuffer[mode] ) {
- for ( int i=0; i<2; i++ ) {
- if ( stream_.userBuffer[i] ) {
- free( stream_.userBuffer[i] );
- stream_.userBuffer[i] = 0;
+ bool makeBuffer = true;
+ bufferBytes = stream_.nDeviceChannels[mode] * formatBytes( stream_.deviceFormat[mode] );
+ if ( mode == INPUT ) {
+ if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) {
+ unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] );
+ if ( bufferBytes <= bytesOut ) makeBuffer = false;
}
}
- if ( stream_.deviceBuffer ) {
- free( stream_.deviceBuffer );
- stream_.deviceBuffer = 0;
+ if ( makeBuffer ) {
+ bufferBytes *= *bufferSize;
+ if ( stream_.deviceBuffer ) free( stream_.deviceBuffer );
+ stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 );
+ if ( stream_.deviceBuffer == NULL ) {
+ errorText_ = "RtApiAsio::probeDeviceOpen: error allocating device buffer memory.";
+ goto error;
+ }
}
-
- stream_.mode = UNINITIALIZED;
- stream_.state = STREAM_CLOSED;
}
- void RtApiAsio :: startStream()
- {
- verifyStream();
- if ( stream_.state == STREAM_RUNNING ) {
- errorText_ = "RtApiAsio::startStream(): the stream is already running!";
- error( RtError::WARNING );
- return;
- }
+ stream_.sampleRate = sampleRate;
+ stream_.device[mode] = device;
+ stream_.state = STREAM_STOPPED;
+ asioCallbackInfo = &stream_.callbackInfo;
+ stream_.callbackInfo.object = (void *) this;
+ if ( stream_.mode == OUTPUT && mode == INPUT )
+ // We had already set up an output stream.
+ stream_.mode = DUPLEX;
+ else
+ stream_.mode = mode;
- MUTEX_LOCK( &stream_.mutex );
+ // Determine device latencies
+ result = ASIOGetLatencies( &inputLatency, &outputLatency );
+ if ( result != ASE_OK ) {
+ errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error (" << getAsioErrorString( result ) << ") getting latency.";
+ errorText_ = errorStream_.str();
+ error( RtError::WARNING); // warn but don't fail
+ }
+ else {
+ stream_.latency[0] = outputLatency;
+ stream_.latency[1] = inputLatency;
+ }
- AsioHandle *handle = (AsioHandle *) stream_.apiHandle;
- ASIOError result = ASIOStart();
- if ( result != ASE_OK ) {
- errorStream_ << "RtApiAsio::startStream: error (" << getAsioErrorString( result ) << ") starting device.";
- errorText_ = errorStream_.str();
- goto unlock;
- }
+ // Setup the buffer conversion information structure. We don't use
+ // buffers to do channel offsets, so we override that parameter
+ // here.
+ if ( stream_.doConvertBuffer[mode] ) setConvertInfo( mode, 0 );
- handle->drainCounter = 0;
- handle->internalDrain = false;
- stream_.state = STREAM_RUNNING;
- asioXRun = false;
+ return SUCCESS;
- unlock:
- MUTEX_UNLOCK( &stream_.mutex );
+ error:
+ if ( buffersAllocated )
+ ASIODisposeBuffers();
+ drivers.removeCurrentDriver();
- if ( result == ASE_OK ) return;
- error( RtError::SYSTEM_ERROR );
+ if ( handle ) {
+ CloseHandle( handle->condition );
+ if ( handle->bufferInfos )
+ free( handle->bufferInfos );
+ delete handle;
+ stream_.apiHandle = 0;
}
- void RtApiAsio :: stopStream()
- {
- verifyStream();
- if ( stream_.state == STREAM_STOPPED ) {
- errorText_ = "RtApiAsio::stopStream(): the stream is already stopped!";
- error( RtError::WARNING );
- return;
+ for ( int i=0; i<2; i++ ) {
+ if ( stream_.userBuffer[i] ) {
+ free( stream_.userBuffer[i] );
+ stream_.userBuffer[i] = 0;
}
+ }
- MUTEX_LOCK( &stream_.mutex );
+ if ( stream_.deviceBuffer ) {
+ free( stream_.deviceBuffer );
+ stream_.deviceBuffer = 0;
+ }
- AsioHandle *handle = (AsioHandle *) stream_.apiHandle;
- if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
- if ( handle->drainCounter == 0 ) {
- handle->drainCounter = 1;
- MUTEX_UNLOCK( &stream_.mutex );
- WaitForMultipleObjects( 1, &handle->condition, FALSE, INFINITE ); // block until signaled
- ResetEvent( handle->condition );
- MUTEX_LOCK( &stream_.mutex );
- }
- }
+ return FAILURE;
+}
- ASIOError result = ASIOStop();
- if ( result != ASE_OK ) {
- errorStream_ << "RtApiAsio::stopStream: error (" << getAsioErrorString( result ) << ") stopping device.";
- errorText_ = errorStream_.str();
- }
+void RtApiAsio :: closeStream()
+{
+ if ( stream_.state == STREAM_CLOSED ) {
+ errorText_ = "RtApiAsio::closeStream(): no open stream to close!";
+ error( RtError::WARNING );
+ return;
+ }
+ if ( stream_.state == STREAM_RUNNING ) {
stream_.state = STREAM_STOPPED;
- MUTEX_UNLOCK( &stream_.mutex );
-
- if ( result == ASE_OK ) return;
- error( RtError::SYSTEM_ERROR );
+ ASIOStop();
}
+ ASIODisposeBuffers();
+ drivers.removeCurrentDriver();
- void RtApiAsio :: abortStream()
- {
- verifyStream();
- if ( stream_.state == STREAM_STOPPED ) {
- errorText_ = "RtApiAsio::abortStream(): the stream is already stopped!";
- error( RtError::WARNING );
- return;
- }
-
- // The following lines were commented-out because some behavior was
- // noted where the device buffers need to be zeroed to avoid
- // continuing sound, even when the device buffers are completed
- // disposed. So now, calling abort is the same as calling stop.
- //AsioHandle *handle = (AsioHandle *) stream_.apiHandle;
- //handle->drainCounter = 1;
- stopStream();
+ AsioHandle *handle = (AsioHandle *) stream_.apiHandle;
+ if ( handle ) {
+ CloseHandle( handle->condition );
+ if ( handle->bufferInfos )
+ free( handle->bufferInfos );
+ delete handle;
+ stream_.apiHandle = 0;
}
- bool RtApiAsio :: callbackEvent( long bufferIndex )
- {
- if ( stream_.state == STREAM_STOPPED ) return SUCCESS;
- if ( stream_.state == STREAM_CLOSED ) {
- errorText_ = "RtApiAsio::callbackEvent(): the stream is closed ... this shouldn't happen!";
- error( RtError::WARNING );
- return FAILURE;
+ for ( int i=0; i<2; i++ ) {
+ if ( stream_.userBuffer[i] ) {
+ free( stream_.userBuffer[i] );
+ stream_.userBuffer[i] = 0;
}
+ }
- CallbackInfo *info = (CallbackInfo *) &stream_.callbackInfo;
- AsioHandle *handle = (AsioHandle *) stream_.apiHandle;
+ if ( stream_.deviceBuffer ) {
+ free( stream_.deviceBuffer );
+ stream_.deviceBuffer = 0;
+ }
- // Check if we were draining the stream and signal is finished.
- if ( handle->drainCounter > 3 ) {
- if ( handle->internalDrain == false )
- SetEvent( handle->condition );
- else
- stopStream();
- return SUCCESS;
- }
+ stream_.mode = UNINITIALIZED;
+ stream_.state = STREAM_CLOSED;
+}
- MUTEX_LOCK( &stream_.mutex );
+void RtApiAsio :: startStream()
+{
+ verifyStream();
+ if ( stream_.state == STREAM_RUNNING ) {
+ errorText_ = "RtApiAsio::startStream(): the stream is already running!";
+ error( RtError::WARNING );
+ return;
+ }
- // The state might change while waiting on a mutex.
- if ( stream_.state == STREAM_STOPPED ) goto unlock;
+ MUTEX_LOCK( &stream_.mutex );
- // Invoke user callback to get fresh output data UNLESS we are
- // draining stream.
- if ( handle->drainCounter == 0 ) {
- RtAudioCallback callback = (RtAudioCallback) info->callback;
- double streamTime = getStreamTime();
- RtAudioStreamStatus status = 0;
- if ( stream_.mode != INPUT && asioXRun == true ) {
- status |= RTAUDIO_OUTPUT_UNDERFLOW;
- asioXRun = false;
- }
- if ( stream_.mode != OUTPUT && asioXRun == true ) {
- status |= RTAUDIO_INPUT_OVERFLOW;
- asioXRun = false;
- }
- handle->drainCounter = callback( stream_.userBuffer[0], stream_.userBuffer[1],
- stream_.bufferSize, streamTime, status, info->userData );
- if ( handle->drainCounter == 2 ) {
- MUTEX_UNLOCK( &stream_.mutex );
- abortStream();
- return SUCCESS;
- }
- else if ( handle->drainCounter == 1 )
- handle->internalDrain = true;
- }
+ AsioHandle *handle = (AsioHandle *) stream_.apiHandle;
+ ASIOError result = ASIOStart();
+ if ( result != ASE_OK ) {
+ errorStream_ << "RtApiAsio::startStream: error (" << getAsioErrorString( result ) << ") starting device.";
+ errorText_ = errorStream_.str();
+ goto unlock;
+ }
- unsigned int nChannels, bufferBytes, i, j;
- nChannels = stream_.nDeviceChannels[0] + stream_.nDeviceChannels[1];
- if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
+ handle->drainCounter = 0;
+ handle->internalDrain = false;
+ stream_.state = STREAM_RUNNING;
+ asioXRun = false;
- bufferBytes = stream_.bufferSize * formatBytes( stream_.deviceFormat[0] );
+ unlock:
+ MUTEX_UNLOCK( &stream_.mutex );
- if ( handle->drainCounter > 1 ) { // write zeros to the output stream
+ if ( result == ASE_OK ) return;
+ error( RtError::SYSTEM_ERROR );
+}
- for ( i=0, j=0; i<nChannels; i++ ) {
- if ( handle->bufferInfos[i].isInput != ASIOTrue )
- memset( handle->bufferInfos[i].buffers[bufferIndex], 0, bufferBytes );
- }
+void RtApiAsio :: stopStream()
+{
+ verifyStream();
+ if ( stream_.state == STREAM_STOPPED ) {
+ errorText_ = "RtApiAsio::stopStream(): the stream is already stopped!";
+ error( RtError::WARNING );
+ return;
+ }
- }
- else if ( stream_.doConvertBuffer[0] ) {
+ MUTEX_LOCK( &stream_.mutex );
- convertBuffer( stream_.deviceBuffer, stream_.userBuffer[0], stream_.convertInfo[0] );
- if ( stream_.doByteSwap[0] )
- byteSwapBuffer( stream_.deviceBuffer,
- stream_.bufferSize * stream_.nDeviceChannels[0],
- stream_.deviceFormat[0] );
-
- for ( i=0, j=0; i<nChannels; i++ ) {
- if ( handle->bufferInfos[i].isInput != ASIOTrue )
- memcpy( handle->bufferInfos[i].buffers[bufferIndex],
- &stream_.deviceBuffer[j++*bufferBytes], bufferBytes );
- }
+ if ( stream_.state == STREAM_STOPPED ) {
+ MUTEX_UNLOCK( &stream_.mutex );
+ return;
+ }
- }
- else {
+ AsioHandle *handle = (AsioHandle *) stream_.apiHandle;
+ if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
+ if ( handle->drainCounter == 0 ) {
+ handle->drainCounter = 1;
+ MUTEX_UNLOCK( &stream_.mutex );
+ WaitForMultipleObjects( 1, &handle->condition, FALSE, INFINITE ); // block until signaled
+ ResetEvent( handle->condition );
+ MUTEX_LOCK( &stream_.mutex );
+ }
+ }
- if ( stream_.doByteSwap[0] )
- byteSwapBuffer( stream_.userBuffer[0],
- stream_.bufferSize * stream_.nUserChannels[0],
- stream_.userFormat );
+ ASIOError result = ASIOStop();
+ if ( result != ASE_OK ) {
+ errorStream_ << "RtApiAsio::stopStream: error (" << getAsioErrorString( result ) << ") stopping device.";
+ errorText_ = errorStream_.str();
+ }
- for ( i=0, j=0; i<nChannels; i++ ) {
- if ( handle->bufferInfos[i].isInput != ASIOTrue )
- memcpy( handle->bufferInfos[i].buffers[bufferIndex],
- &stream_.userBuffer[0][bufferBytes*j++], bufferBytes );
- }
+ stream_.state = STREAM_STOPPED;
+ MUTEX_UNLOCK( &stream_.mutex );
- }
+ if ( result == ASE_OK ) return;
+ error( RtError::SYSTEM_ERROR );
+}
- if ( handle->drainCounter ) {
- handle->drainCounter++;
- goto unlock;
- }
- }
+void RtApiAsio :: abortStream()
+{
+ verifyStream();
+ if ( stream_.state == STREAM_STOPPED ) {
+ errorText_ = "RtApiAsio::abortStream(): the stream is already stopped!";
+ error( RtError::WARNING );
+ return;
+ }
+
+ // The following lines were commented-out because some behavior was
+ // noted where the device buffers need to be zeroed to avoid
+ // continuing sound, even when the device buffers are completely
+ // disposed. So now, calling abort is the same as calling stop.
+ // AsioHandle *handle = (AsioHandle *) stream_.apiHandle;
+ // handle->drainCounter = 1;
+ stopStream();
+}
+
+bool RtApiAsio :: callbackEvent( long bufferIndex )
+{
+ if ( stream_.state == STREAM_STOPPED ) return SUCCESS;
+ if ( stream_.state == STREAM_CLOSED ) {
+ errorText_ = "RtApiAsio::callbackEvent(): the stream is closed ... this shouldn't happen!";
+ error( RtError::WARNING );
+ return FAILURE;
+ }
+
+ CallbackInfo *info = (CallbackInfo *) &stream_.callbackInfo;
+ AsioHandle *handle = (AsioHandle *) stream_.apiHandle;
- if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {
+ // Check if we were draining the stream and signal is finished.
+ if ( handle->drainCounter > 3 ) {
+ if ( handle->internalDrain == false )
+ SetEvent( handle->condition );
+ else
+ stopStream();
+ return SUCCESS;
+ }
- bufferBytes = stream_.bufferSize * formatBytes(stream_.deviceFormat[1]);
+ MUTEX_LOCK( &stream_.mutex );
- if (stream_.doConvertBuffer[1]) {
+ // The state might change while waiting on a mutex.
+ if ( stream_.state == STREAM_STOPPED ) goto unlock;
- // Always interleave ASIO input data.
- for ( i=0, j=0; i<nChannels; i++ ) {
- if ( handle->bufferInfos[i].isInput == ASIOTrue )
- memcpy( &stream_.deviceBuffer[j++*bufferBytes],
- handle->bufferInfos[i].buffers[bufferIndex],
- bufferBytes );
- }
+ // Invoke user callback to get fresh output data UNLESS we are
+ // draining stream.
+ if ( handle->drainCounter == 0 ) {
+ RtAudioCallback callback = (RtAudioCallback) info->callback;
+ double streamTime = getStreamTime();
+ RtAudioStreamStatus status = 0;
+ if ( stream_.mode != INPUT && asioXRun == true ) {
+ status |= RTAUDIO_OUTPUT_UNDERFLOW;
+ asioXRun = false;
+ }
+ if ( stream_.mode != OUTPUT && asioXRun == true ) {
+ status |= RTAUDIO_INPUT_OVERFLOW;
+ asioXRun = false;
+ }
+ handle->drainCounter = callback( stream_.userBuffer[0], stream_.userBuffer[1],
+ stream_.bufferSize, streamTime, status, info->userData );
+ if ( handle->drainCounter == 2 ) {
+ MUTEX_UNLOCK( &stream_.mutex );
+ abortStream();
+ return SUCCESS;
+ }
+ else if ( handle->drainCounter == 1 )
+ handle->internalDrain = true;
+ }
+
+ 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] );
- if ( stream_.doByteSwap[1] )
- byteSwapBuffer( stream_.deviceBuffer,
- stream_.bufferSize * stream_.nDeviceChannels[1],
- stream_.deviceFormat[1] );
- convertBuffer( stream_.userBuffer[1], stream_.deviceBuffer, stream_.convertInfo[1] );
+ if ( handle->drainCounter > 1 ) { // write zeros to the output stream
+ for ( i=0, j=0; i<nChannels; i++ ) {
+ if ( handle->bufferInfos[i].isInput != ASIOTrue )
+ memset( handle->bufferInfos[i].buffers[bufferIndex], 0, bufferBytes );
}
- else {
- for ( i=0, j=0; i<nChannels; i++ ) {
- if ( handle->bufferInfos[i].isInput == ASIOTrue ) {
- memcpy( &stream_.userBuffer[1][bufferBytes*j++],
- handle->bufferInfos[i].buffers[bufferIndex],
- bufferBytes );
- }
- }
- if ( stream_.doByteSwap[1] )
- byteSwapBuffer( stream_.userBuffer[1],
- stream_.bufferSize * stream_.nUserChannels[1],
- stream_.userFormat );
+ }
+ else if ( stream_.doConvertBuffer[0] ) {
+
+ convertBuffer( stream_.deviceBuffer, stream_.userBuffer[0], stream_.convertInfo[0] );
+ if ( stream_.doByteSwap[0] )
+ byteSwapBuffer( stream_.deviceBuffer,
+ stream_.bufferSize * stream_.nDeviceChannels[0],
+ stream_.deviceFormat[0] );
+
+ for ( i=0, j=0; i<nChannels; i++ ) {
+ if ( handle->bufferInfos[i].isInput != ASIOTrue )
+ memcpy( handle->bufferInfos[i].buffers[bufferIndex],
+ &stream_.deviceBuffer[j++*bufferBytes], bufferBytes );
}
+
}
+ else {
- unlock:
- // The following call was suggested by Malte Clasen. While the API
- // documentation indicates it should not be required, some device
- // drivers apparently do not function correctly without it.
- ASIOOutputReady();
+ if ( stream_.doByteSwap[0] )
+ byteSwapBuffer( stream_.userBuffer[0],
+ stream_.bufferSize * stream_.nUserChannels[0],
+ stream_.userFormat );
- MUTEX_UNLOCK( &stream_.mutex );
+ for ( i=0, j=0; i<nChannels; i++ ) {
+ if ( handle->bufferInfos[i].isInput != ASIOTrue )
+ memcpy( handle->bufferInfos[i].buffers[bufferIndex],
+ &stream_.userBuffer[0][bufferBytes*j++], bufferBytes );
+ }
- RtApi::tickStreamTime();
- return SUCCESS;
+ }
+
+ if ( handle->drainCounter ) {
+ handle->drainCounter++;
+ goto unlock;
+ }
}
- void sampleRateChanged( ASIOSampleRate sRate )
- {
- // The ASIO documentation says that this usually only happens during
- // external sync. Audio processing is not stopped by the driver,
- // actual sample rate might not have even changed, maybe only the
- // sample rate status of an AES/EBU or S/PDIF digital input at the
- // audio device.
+ if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {
+
+ bufferBytes = stream_.bufferSize * formatBytes(stream_.deviceFormat[1]);
+
+ if (stream_.doConvertBuffer[1]) {
+
+ // Always interleave ASIO input data.
+ for ( i=0, j=0; i<nChannels; i++ ) {
+ if ( handle->bufferInfos[i].isInput == ASIOTrue )
+ memcpy( &stream_.deviceBuffer[j++*bufferBytes],
+ handle->bufferInfos[i].buffers[bufferIndex],
+ bufferBytes );
+ }
+
+ if ( stream_.doByteSwap[1] )
+ byteSwapBuffer( stream_.deviceBuffer,
+ stream_.bufferSize * stream_.nDeviceChannels[1],
+ stream_.deviceFormat[1] );
+ convertBuffer( stream_.userBuffer[1], stream_.deviceBuffer, stream_.convertInfo[1] );
- RtApi *object = (RtApi *) asioCallbackInfo->object;
- try {
- object->stopStream();
}
- catch ( RtError &exception ) {
- std::cerr << "\nRtApiAsio: sampleRateChanged() error (" << exception.getMessage() << ")!\n" << std::endl;
- return;
+ else {
+ for ( i=0, j=0; i<nChannels; i++ ) {
+ if ( handle->bufferInfos[i].isInput == ASIOTrue ) {
+ memcpy( &stream_.userBuffer[1][bufferBytes*j++],
+ handle->bufferInfos[i].buffers[bufferIndex],
+ bufferBytes );
+ }
+ }
+
+ if ( stream_.doByteSwap[1] )
+ byteSwapBuffer( stream_.userBuffer[1],
+ stream_.bufferSize * stream_.nUserChannels[1],
+ stream_.userFormat );
}
+ }
+
+ unlock:
+ // The following call was suggested by Malte Clasen. While the API
+ // documentation indicates it should not be required, some device
+ // drivers apparently do not function correctly without it.
+ ASIOOutputReady();
+
+ MUTEX_UNLOCK( &stream_.mutex );
+
+ RtApi::tickStreamTime();
+ return SUCCESS;
+}
- std::cerr << "\nRtApiAsio: driver reports sample rate changed to " << sRate << " ... stream stopped!!!\n" << std::endl;
+void sampleRateChanged( ASIOSampleRate sRate )
+{
+ // The ASIO documentation says that this usually only happens during
+ // external sync. Audio processing is not stopped by the driver,
+ // actual sample rate might not have even changed, maybe only the
+ // sample rate status of an AES/EBU or S/PDIF digital input at the
+ // audio device.
+
+ RtApi *object = (RtApi *) asioCallbackInfo->object;
+ try {
+ object->stopStream();
+ }
+ catch ( RtError &exception ) {
+ std::cerr << "\nRtApiAsio: sampleRateChanged() error (" << exception.getMessage() << ")!\n" << std::endl;
+ return;
}
- long asioMessages( long selector, long value, void* message, double* opt )
- {
- long ret = 0;
-
- switch( selector ) {
- case kAsioSelectorSupported:
- if ( value == kAsioResetRequest
- || value == kAsioEngineVersion
- || value == kAsioResyncRequest
- || value == kAsioLatenciesChanged
- // The following three were added for ASIO 2.0, you don't
- // necessarily have to support them.
- || value == kAsioSupportsTimeInfo
- || value == kAsioSupportsTimeCode
- || value == kAsioSupportsInputMonitor)
- ret = 1L;
- break;
- case kAsioResetRequest:
- // Defer the task and perform the reset of the driver during the
- // next "safe" situation. You cannot reset the driver right now,
- // as this code is called from the driver. Reset the driver is
- // done by completely destruct is. I.e. ASIOStop(),
- // ASIODisposeBuffers(), Destruction Afterwards you initialize the
- // driver again.
- std::cerr << "\nRtApiAsio: driver reset requested!!!" << std::endl;
- ret = 1L;
- break;
- case kAsioResyncRequest:
- // This informs the application that the driver encountered some
- // non-fatal data loss. It is used for synchronization purposes
- // of different media. Added mainly to work around the Win16Mutex
- // problems in Windows 95/98 with the Windows Multimedia system,
- // which could lose data because the Mutex was held too long by
- // another thread. However a driver can issue it in other
- // situations, too.
- // std::cerr << "\nRtApiAsio: driver resync requested!!!" << std::endl;
- asioXRun = true;
- ret = 1L;
- break;
- case kAsioLatenciesChanged:
- // This will inform the host application that the drivers were
- // latencies changed. Beware, it this does not mean that the
- // buffer sizes have changed! You might need to update internal
- // delay data.
- std::cerr << "\nRtApiAsio: driver latency may have changed!!!" << std::endl;
+ std::cerr << "\nRtApiAsio: driver reports sample rate changed to " << sRate << " ... stream stopped!!!\n" << std::endl;
+}
+
+long asioMessages( long selector, long value, void* message, double* opt )
+{
+ long ret = 0;
+
+ switch( selector ) {
+ case kAsioSelectorSupported:
+ if ( value == kAsioResetRequest
+ || value == kAsioEngineVersion
+ || value == kAsioResyncRequest
+ || value == kAsioLatenciesChanged
+ // The following three were added for ASIO 2.0, you don't
+ // necessarily have to support them.
+ || value == kAsioSupportsTimeInfo
+ || value == kAsioSupportsTimeCode
+ || value == kAsioSupportsInputMonitor)
ret = 1L;
- break;
- case kAsioEngineVersion:
- // Return the supported ASIO version of the host application. If
- // a host application does not implement this selector, ASIO 1.0
- // is assumed by the driver.
- ret = 2L;
- break;
- case kAsioSupportsTimeInfo:
- // Informs the driver whether the
- // asioCallbacks.bufferSwitchTimeInfo() callback is supported.
- // For compatibility with ASIO 1.0 drivers the host application
- // should always support the "old" bufferSwitch method, too.
- ret = 0;
- break;
- case kAsioSupportsTimeCode:
- // Informs the driver whether application is interested in time
- // code info. If an application does not need to know about time
- // code, the driver has less work to do.
- ret = 0;
- break;
- }
- return ret;
- }
+ break;
+ case kAsioResetRequest:
+ // Defer the task and perform the reset of the driver during the
+ // next "safe" situation. You cannot reset the driver right now,
+ // as this code is called from the driver. Reset the driver is
+ // done by completely destruct is. I.e. ASIOStop(),
+ // ASIODisposeBuffers(), Destruction Afterwards you initialize the
+ // driver again.
+ std::cerr << "\nRtApiAsio: driver reset requested!!!" << std::endl;
+ ret = 1L;
+ break;
+ case kAsioResyncRequest:
+ // This informs the application that the driver encountered some
+ // non-fatal data loss. It is used for synchronization purposes
+ // of different media. Added mainly to work around the Win16Mutex
+ // problems in Windows 95/98 with the Windows Multimedia system,
+ // which could lose data because the Mutex was held too long by
+ // another thread. However a driver can issue it in other
+ // situations, too.
+ // std::cerr << "\nRtApiAsio: driver resync requested!!!" << std::endl;
+ asioXRun = true;
+ ret = 1L;
+ break;
+ case kAsioLatenciesChanged:
+ // This will inform the host application that the drivers were
+ // latencies changed. Beware, it this does not mean that the
+ // buffer sizes have changed! You might need to update internal
+ // delay data.
+ std::cerr << "\nRtApiAsio: driver latency may have changed!!!" << std::endl;
+ ret = 1L;
+ break;
+ case kAsioEngineVersion:
+ // Return the supported ASIO version of the host application. If
+ // a host application does not implement this selector, ASIO 1.0
+ // is assumed by the driver.
+ ret = 2L;
+ break;
+ case kAsioSupportsTimeInfo:
+ // Informs the driver whether the
+ // asioCallbacks.bufferSwitchTimeInfo() callback is supported.
+ // For compatibility with ASIO 1.0 drivers the host application
+ // should always support the "old" bufferSwitch method, too.
+ ret = 0;
+ break;
+ case kAsioSupportsTimeCode:
+ // Informs the driver whether application is interested in time
+ // code info. If an application does not need to know about time
+ // code, the driver has less work to do.
+ ret = 0;
+ break;
+ }
+ return ret;
+}
- static const char* getAsioErrorString( ASIOError result )
+static const char* getAsioErrorString( ASIOError result )
+{
+ struct Messages
{
- struct Messages
+ ASIOError value;
+ const char*message;
+ };
+
+ static Messages m[] =
{
- ASIOError value;
- const char*message;
+ { ASE_NotPresent, "Hardware input or output is not present or available." },
+ { ASE_HWMalfunction, "Hardware is malfunctioning." },
+ { ASE_InvalidParameter, "Invalid input parameter." },
+ { ASE_InvalidMode, "Invalid mode." },
+ { ASE_SPNotAdvancing, "Sample position not advancing." },
+ { ASE_NoClock, "Sample clock or rate cannot be determined or is not present." },
+ { ASE_NoMemory, "Not enough memory to complete the request." }
};
- static Messages m[] =
- {
- { ASE_NotPresent, "Hardware input or output is not present or available." },
- { ASE_HWMalfunction, "Hardware is malfunctioning." },
- { ASE_InvalidParameter, "Invalid input parameter." },
- { ASE_InvalidMode, "Invalid mode." },
- { ASE_SPNotAdvancing, "Sample position not advancing." },
- { ASE_NoClock, "Sample clock or rate cannot be determined or is not present." },
- { ASE_NoMemory, "Not enough memory to complete the request." }
- };
+ for ( unsigned int i = 0; i < sizeof(m)/sizeof(m[0]); ++i )
+ if ( m[i].value == result ) return m[i].message;
- for ( unsigned int i = 0; i < sizeof(m)/sizeof(m[0]); ++i )
- if ( m[i].value == result ) return m[i].message;
-
- return "Unknown error.";
- }
- //******************** End of __WINDOWS_ASIO__ *********************//
+ return "Unknown error.";
+}
+//******************** End of __WINDOWS_ASIO__ *********************//
#endif
#if defined(__WINDOWS_DS__) // Windows DirectSound API
- // Modified by Robin Davies, October 2005
- // - Improvements to DirectX pointer chasing.
- // - Backdoor RtDsStatistics hook provides DirectX performance information.
- // - Bug fix for non-power-of-two Asio granularity used by Edirol PCR-A30.
- // - Auto-call CoInitialize for DSOUND and ASIO platforms.
- // Various revisions for RtAudio 4.0 by Gary Scavone, April 2007
+// Modified by Robin Davies, October 2005
+// - Improvements to DirectX pointer chasing.
+// - Bug fix for non-power-of-two Asio granularity used by Edirol PCR-A30.
+// - Auto-call CoInitialize for DSOUND and ASIO platforms.
+// Various revisions for RtAudio 4.0 by Gary Scavone, April 2007
+// Changed device query structure for RtAudio 4.0.7, January 2010
#include <dsound.h>
#include <assert.h>
+#include <algorithm>
#if defined(__MINGW32__)
// missing from latest mingw winapi
#pragma comment( lib, "winmm.lib" ) // then, auto-link winmm.lib. Otherwise, it has to be added manually.
#endif
- static inline DWORD dsPointerDifference( DWORD laterPointer, DWORD earlierPointer, DWORD bufferSize )
- {
- if ( laterPointer > earlierPointer )
- return laterPointer - earlierPointer;
- else
- return laterPointer - earlierPointer + bufferSize;
- }
+static inline DWORD dsPointerBetween( DWORD pointer, DWORD laterPointer, DWORD earlierPointer, DWORD bufferSize )
+{
+ if ( pointer > bufferSize ) pointer -= bufferSize;
+ if ( laterPointer < earlierPointer ) laterPointer += bufferSize;
+ if ( pointer < earlierPointer ) pointer += bufferSize;
+ return pointer >= earlierPointer && pointer < laterPointer;
+}
- static inline DWORD dsPointerBetween( DWORD pointer, DWORD laterPointer, DWORD earlierPointer, DWORD bufferSize )
- {
- if ( pointer > bufferSize ) pointer -= bufferSize;
- if ( laterPointer < earlierPointer ) laterPointer += bufferSize;
- if ( pointer < earlierPointer ) pointer += bufferSize;
- return pointer >= earlierPointer && pointer < laterPointer;
- }
-
- // A structure to hold various information related to the DirectSound
- // API implementation.
- struct DsHandle {
- unsigned int drainCounter; // Tracks callback counts when draining
- bool internalDrain; // Indicates if stop is initiated from callback or not.
- void *id[2];
- void *buffer[2];
- bool xrun[2];
- UINT bufferPointer[2];
- DWORD dsBufferSize[2];
- DWORD dsPointerLeadTime[2]; // the number of bytes ahead of the safe pointer to lead by.
- HANDLE condition;
-
- DsHandle()
- :drainCounter(0), internalDrain(false) { id[0] = 0; id[1] = 0; buffer[0] = 0; buffer[1] = 0; xrun[0] = false; xrun[1] = false; bufferPointer[0] = 0; bufferPointer[1] = 0; }
- };
+// A structure to hold various information related to the DirectSound
+// API implementation.
+struct DsHandle {
+ unsigned int drainCounter; // Tracks callback counts when draining
+ bool internalDrain; // Indicates if stop is initiated from callback or not.
+ void *id[2];
+ void *buffer[2];
+ bool xrun[2];
+ UINT bufferPointer[2];
+ DWORD dsBufferSize[2];
+ DWORD dsPointerLeadTime[2]; // the number of bytes ahead of the safe pointer to lead by.
+ HANDLE condition;
- /*
- RtApiDs::RtDsStatistics RtApiDs::statistics;
+ DsHandle()
+ :drainCounter(0), internalDrain(false) { id[0] = 0; id[1] = 0; buffer[0] = 0; buffer[1] = 0; xrun[0] = false; xrun[1] = false; bufferPointer[0] = 0; bufferPointer[1] = 0; }
+};
- // Provides a backdoor hook to monitor for DirectSound read overruns and write underruns.
- RtApiDs::RtDsStatistics RtApiDs::getDsStatistics()
- {
- RtDsStatistics s = statistics;
+// Declarations for utility functions, callbacks, and structures
+// specific to the DirectSound implementation.
+static BOOL CALLBACK deviceQueryCallback( LPGUID lpguid,
+ LPCTSTR description,
+ LPCTSTR module,
+ LPVOID lpContext );
- // update the calculated fields.
- if ( s.inputFrameSize != 0 )
- s.latency += s.readDeviceSafeLeadBytes * 1.0 / s.inputFrameSize / s.sampleRate;
+static const char* getErrorString( int code );
- if ( s.outputFrameSize != 0 )
- s.latency += (s.writeDeviceSafeLeadBytes + s.writeDeviceBufferLeadBytes) * 1.0 / s.outputFrameSize / s.sampleRate;
+extern "C" unsigned __stdcall callbackHandler( void *ptr );
- return s;
- }
- */
+struct DsDevice {
+ LPGUID id[2];
+ bool validId[2];
+ bool found;
+ std::string name;
- // Declarations for utility functions, callbacks, and structures
- // specific to the DirectSound implementation.
- static BOOL CALLBACK deviceQueryCallback( LPGUID lpguid,
- LPCTSTR description,
- LPCTSTR module,
- LPVOID lpContext );
+ DsDevice()
+ : found(false) { validId[0] = false; validId[1] = false; }
+};
- static char* getErrorString( int code );
+std::vector< DsDevice > dsDevices;
- extern "C" unsigned __stdcall callbackHandler( void *ptr );
+RtApiDs :: RtApiDs()
+{
+ // Dsound will run both-threaded. If CoInitialize fails, then just
+ // accept whatever the mainline chose for a threading model.
+ coInitialized_ = false;
+ HRESULT hr = CoInitialize( NULL );
+ if ( !FAILED( hr ) ) coInitialized_ = true;
+}
- struct EnumInfo {
- bool isInput;
- bool getDefault;
- bool findIndex;
- unsigned int counter;
- unsigned int index;
- LPGUID id;
- std::string name;
+RtApiDs :: ~RtApiDs()
+{
+ if ( coInitialized_ ) CoUninitialize(); // balanced call.
+ if ( stream_.state != STREAM_CLOSED ) closeStream();
+}
- EnumInfo()
- : isInput(false), getDefault(false), findIndex(false), counter(0), index(0) {}
- };
+// The DirectSound default output is always the first device.
+unsigned int RtApiDs :: getDefaultOutputDevice( void )
+{
+ return 0;
+}
- RtApiDs :: RtApiDs()
- {
- // Dsound will run both-threaded. If CoInitialize fails, then just
- // accept whatever the mainline chose for a threading model.
- coInitialized_ = false;
- HRESULT hr = CoInitialize( NULL );
- if ( !FAILED( hr ) ) coInitialized_ = true;
+// The DirectSound default input is always the first input device,
+// which is the first capture device enumerated.
+unsigned int RtApiDs :: getDefaultInputDevice( void )
+{
+ return 0;
+}
+
+unsigned int RtApiDs :: getDeviceCount( void )
+{
+ // Set query flag for previously found devices to false, so that we
+ // can check for any devices that have disappeared.
+ for ( unsigned int i=0; i<dsDevices.size(); i++ )
+ dsDevices[i].found = false;
+
+ // Query DirectSound devices.
+ bool isInput = false;
+ HRESULT result = DirectSoundEnumerate( (LPDSENUMCALLBACK) deviceQueryCallback, &isInput );
+ if ( FAILED( result ) ) {
+ errorStream_ << "RtApiDs::getDeviceCount: error (" << getErrorString( result ) << ") enumerating output devices!";
+ errorText_ = errorStream_.str();
+ error( RtError::WARNING );
}
- RtApiDs :: ~RtApiDs()
- {
- if ( coInitialized_ ) CoUninitialize(); // balanced call.
- if ( stream_.state != STREAM_CLOSED ) closeStream();
+ // Query DirectSoundCapture devices.
+ isInput = true;
+ result = DirectSoundCaptureEnumerate( (LPDSENUMCALLBACK) deviceQueryCallback, &isInput );
+ if ( FAILED( result ) ) {
+ errorStream_ << "RtApiDs::getDeviceCount: error (" << getErrorString( result ) << ") enumerating input devices!";
+ errorText_ = errorStream_.str();
+ error( RtError::WARNING );
}
- unsigned int RtApiDs :: getDefaultInputDevice( void )
- {
- // Count output devices.
- EnumInfo info;
- HRESULT result = DirectSoundEnumerate( (LPDSENUMCALLBACK) deviceQueryCallback, &info );
- if ( FAILED( result ) ) {
- errorStream_ << "RtApiDs::getDefaultOutputDevice: error (" << getErrorString( result ) << ") counting output devices!";
- errorText_ = errorStream_.str();
- error( RtError::WARNING );
- return 0;
- }
+ // Clean out any devices that may have disappeared.
+ std::vector< DsDevice > :: iterator it;
+ for ( it=dsDevices.begin(); it < dsDevices.end(); it++ )
+ if ( it->found == false ) dsDevices.erase( it );
- // Now enumerate input devices until we find the id = NULL.
- info.isInput = true;
- info.getDefault = true;
- result = DirectSoundCaptureEnumerate( (LPDSENUMCALLBACK) deviceQueryCallback, &info );
- if ( FAILED( result ) ) {
- errorStream_ << "RtApiDs::getDefaultInputDevice: error (" << getErrorString( result ) << ") enumerating input devices!";
- errorText_ = errorStream_.str();
- error( RtError::WARNING );
- return 0;
+ return dsDevices.size();
+}
+
+RtAudio::DeviceInfo RtApiDs :: getDeviceInfo( unsigned int device )
+{
+ RtAudio::DeviceInfo info;
+ info.probed = false;
+
+ if ( dsDevices.size() == 0 ) {
+ // Force a query of all devices
+ getDeviceCount();
+ if ( dsDevices.size() == 0 ) {
+ errorText_ = "RtApiDs::getDeviceInfo: no devices found!";
+ error( RtError::INVALID_USE );
}
+ }
- if ( info.counter > 0 ) return info.counter - 1;
- return 0;
+ if ( device >= dsDevices.size() ) {
+ errorText_ = "RtApiDs::getDeviceInfo: device ID is invalid!";
+ error( RtError::INVALID_USE );
}
- unsigned int RtApiDs :: getDefaultOutputDevice( void )
- {
- // Enumerate output devices until we find the id = NULL.
- EnumInfo info;
- info.getDefault = true;
- HRESULT result = DirectSoundEnumerate( (LPDSENUMCALLBACK) deviceQueryCallback, &info );
- if ( FAILED( result ) ) {
- errorStream_ << "RtApiDs::getDefaultOutputDevice: error (" << getErrorString( result ) << ") enumerating output devices!";
- errorText_ = errorStream_.str();
- error( RtError::WARNING );
- return 0;
- }
+ HRESULT result;
+ if ( dsDevices[ device ].validId[0] == false ) goto probeInput;
- if ( info.counter > 0 ) return info.counter - 1;
- return 0;
+ LPDIRECTSOUND output;
+ DSCAPS outCaps;
+ result = DirectSoundCreate( dsDevices[ device ].id[0], &output, NULL );
+ if ( FAILED( result ) ) {
+ errorStream_ << "RtApiDs::getDeviceInfo: error (" << getErrorString( result ) << ") opening output device (" << dsDevices[ device ].name << ")!";
+ errorText_ = errorStream_.str();
+ error( RtError::WARNING );
+ goto probeInput;
}
- unsigned int RtApiDs :: getDeviceCount( void )
- {
- // Count DirectSound devices.
- EnumInfo info;
- HRESULT result = DirectSoundEnumerate( (LPDSENUMCALLBACK) deviceQueryCallback, &info );
- if ( FAILED( result ) ) {
- errorStream_ << "RtApiDs::getDeviceCount: error (" << getErrorString( result ) << ") enumerating output devices!";
- errorText_ = errorStream_.str();
- error( RtError::WARNING );
- }
+ outCaps.dwSize = sizeof( outCaps );
+ result = output->GetCaps( &outCaps );
+ if ( FAILED( result ) ) {
+ output->Release();
+ errorStream_ << "RtApiDs::getDeviceInfo: error (" << getErrorString( result ) << ") getting capabilities!";
+ errorText_ = errorStream_.str();
+ error( RtError::WARNING );
+ goto probeInput;
+ }
- // Count DirectSoundCapture devices.
- info.isInput = true;
- result = DirectSoundCaptureEnumerate( (LPDSENUMCALLBACK) deviceQueryCallback, &info );
- if ( FAILED( result ) ) {
- errorStream_ << "RtApiDs::getDeviceCount: error (" << getErrorString( result ) << ") enumerating input devices!";
- errorText_ = errorStream_.str();
- error( RtError::WARNING );
- }
+ // Get output channel information.
+ info.outputChannels = ( outCaps.dwFlags & DSCAPS_PRIMARYSTEREO ) ? 2 : 1;
- return info.counter;
+ // Get sample rate information.
+ info.sampleRates.clear();
+ for ( unsigned int k=0; k<MAX_SAMPLE_RATES; k++ ) {
+ if ( SAMPLE_RATES[k] >= (unsigned int) outCaps.dwMinSecondarySampleRate &&
+ SAMPLE_RATES[k] <= (unsigned int) outCaps.dwMaxSecondarySampleRate )
+ info.sampleRates.push_back( SAMPLE_RATES[k] );
}
- RtAudio::DeviceInfo RtApiDs :: getDeviceInfo( unsigned int device )
- {
- // Because DirectSound always enumerates input and output devices
- // separately (and because we don't attempt to combine devices
- // internally), none of our "devices" will ever be duplex.
-
- RtAudio::DeviceInfo info;
- info.probed = false;
-
- // Enumerate through devices to find the id (if it exists). Note
- // that we have to do the output enumeration first, even if this is
- // an input device, in order for the device counter to be correct.
- EnumInfo dsinfo;
- dsinfo.findIndex = true;
- dsinfo.index = device;
- HRESULT result = DirectSoundEnumerate( (LPDSENUMCALLBACK) deviceQueryCallback, &dsinfo );
- if ( FAILED( result ) ) {
- errorStream_ << "RtApiDs::getDeviceInfo: error (" << getErrorString( result ) << ") enumerating output devices!";
+ // Get format information.
+ if ( outCaps.dwFlags & DSCAPS_PRIMARY16BIT ) info.nativeFormats |= RTAUDIO_SINT16;
+ if ( outCaps.dwFlags & DSCAPS_PRIMARY8BIT ) info.nativeFormats |= RTAUDIO_SINT8;
+
+ output->Release();
+
+ if ( getDefaultOutputDevice() == device )
+ info.isDefaultOutput = true;
+
+ if ( dsDevices[ device ].validId[1] == false ) {
+ info.name = dsDevices[ device ].name;
+ info.probed = true;
+ return info;
+ }
+
+ probeInput:
+
+ LPDIRECTSOUNDCAPTURE input;
+ result = DirectSoundCaptureCreate( dsDevices[ device ].id[1], &input, NULL );
+ if ( FAILED( result ) ) {
+ errorStream_ << "RtApiDs::getDeviceInfo: error (" << getErrorString( result ) << ") opening input device (" << dsDevices[ device ].name << ")!";
+ errorText_ = errorStream_.str();
+ error( RtError::WARNING );
+ return info;
+ }
+
+ DSCCAPS inCaps;
+ inCaps.dwSize = sizeof( inCaps );
+ result = input->GetCaps( &inCaps );
+ if ( FAILED( result ) ) {
+ input->Release();
+ errorStream_ << "RtApiDs::getDeviceInfo: error (" << getErrorString( result ) << ") getting object capabilities (" << dsDevices[ device ].name << ")!";
+ errorText_ = errorStream_.str();
+ error( RtError::WARNING );
+ return info;
+ }
+
+ // Get input channel information.
+ info.inputChannels = inCaps.dwChannels;
+
+ // Get sample rate and format information.
+ std::vector<unsigned int> rates;
+ if ( inCaps.dwChannels == 2 ) {
+ if ( inCaps.dwFormats & WAVE_FORMAT_1S16 ) info.nativeFormats |= RTAUDIO_SINT16;
+ if ( inCaps.dwFormats & WAVE_FORMAT_2S16 ) info.nativeFormats |= RTAUDIO_SINT16;
+ if ( inCaps.dwFormats & WAVE_FORMAT_4S16 ) info.nativeFormats |= RTAUDIO_SINT16;
+ if ( inCaps.dwFormats & WAVE_FORMAT_96S16 ) info.nativeFormats |= RTAUDIO_SINT16;
+ if ( inCaps.dwFormats & WAVE_FORMAT_1S08 ) info.nativeFormats |= RTAUDIO_SINT8;
+ if ( inCaps.dwFormats & WAVE_FORMAT_2S08 ) info.nativeFormats |= RTAUDIO_SINT8;
+ if ( inCaps.dwFormats & WAVE_FORMAT_4S08 ) info.nativeFormats |= RTAUDIO_SINT8;
+ if ( inCaps.dwFormats & WAVE_FORMAT_96S08 ) info.nativeFormats |= RTAUDIO_SINT8;
+
+ if ( info.nativeFormats & RTAUDIO_SINT16 ) {
+ if ( inCaps.dwFormats & WAVE_FORMAT_1S16 ) rates.push_back( 11025 );
+ if ( inCaps.dwFormats & WAVE_FORMAT_2S16 ) rates.push_back( 22050 );
+ if ( inCaps.dwFormats & WAVE_FORMAT_4S16 ) rates.push_back( 44100 );
+ if ( inCaps.dwFormats & WAVE_FORMAT_96S16 ) rates.push_back( 96000 );
+ }
+ else if ( info.nativeFormats & RTAUDIO_SINT8 ) {
+ if ( inCaps.dwFormats & WAVE_FORMAT_1S08 ) rates.push_back( 11025 );
+ if ( inCaps.dwFormats & WAVE_FORMAT_2S08 ) rates.push_back( 22050 );
+ if ( inCaps.dwFormats & WAVE_FORMAT_4S08 ) rates.push_back( 44100 );
+ if ( inCaps.dwFormats & WAVE_FORMAT_96S08 ) rates.push_back( 96000 );
+ }
+ }
+ else if ( inCaps.dwChannels == 1 ) {
+ if ( inCaps.dwFormats & WAVE_FORMAT_1M16 ) info.nativeFormats |= RTAUDIO_SINT16;
+ if ( inCaps.dwFormats & WAVE_FORMAT_2M16 ) info.nativeFormats |= RTAUDIO_SINT16;
+ if ( inCaps.dwFormats & WAVE_FORMAT_4M16 ) info.nativeFormats |= RTAUDIO_SINT16;
+ if ( inCaps.dwFormats & WAVE_FORMAT_96M16 ) info.nativeFormats |= RTAUDIO_SINT16;
+ if ( inCaps.dwFormats & WAVE_FORMAT_1M08 ) info.nativeFormats |= RTAUDIO_SINT8;
+ if ( inCaps.dwFormats & WAVE_FORMAT_2M08 ) info.nativeFormats |= RTAUDIO_SINT8;
+ if ( inCaps.dwFormats & WAVE_FORMAT_4M08 ) info.nativeFormats |= RTAUDIO_SINT8;
+ if ( inCaps.dwFormats & WAVE_FORMAT_96M08 ) info.nativeFormats |= RTAUDIO_SINT8;
+
+ if ( info.nativeFormats & RTAUDIO_SINT16 ) {
+ if ( inCaps.dwFormats & WAVE_FORMAT_1M16 ) rates.push_back( 11025 );
+ if ( inCaps.dwFormats & WAVE_FORMAT_2M16 ) rates.push_back( 22050 );
+ if ( inCaps.dwFormats & WAVE_FORMAT_4M16 ) rates.push_back( 44100 );
+ if ( inCaps.dwFormats & WAVE_FORMAT_96M16 ) rates.push_back( 96000 );
+ }
+ else if ( info.nativeFormats & RTAUDIO_SINT8 ) {
+ if ( inCaps.dwFormats & WAVE_FORMAT_1M08 ) rates.push_back( 11025 );
+ if ( inCaps.dwFormats & WAVE_FORMAT_2M08 ) rates.push_back( 22050 );
+ if ( inCaps.dwFormats & WAVE_FORMAT_4M08 ) rates.push_back( 44100 );
+ if ( inCaps.dwFormats & WAVE_FORMAT_96M08 ) rates.push_back( 96000 );
+ }
+ }
+ else info.inputChannels = 0; // technically, this would be an error
+
+ input->Release();
+
+ if ( info.inputChannels == 0 ) return info;
+
+ // Copy the supported rates to the info structure but avoid duplication.
+ bool found;
+ for ( unsigned int i=0; i<rates.size(); i++ ) {
+ found = false;
+ for ( unsigned int j=0; j<info.sampleRates.size(); j++ ) {
+ if ( rates[i] == info.sampleRates[j] ) {
+ found = true;
+ break;
+ }
+ }
+ if ( found == false ) info.sampleRates.push_back( rates[i] );
+ }
+ sort( info.sampleRates.begin(), info.sampleRates.end() );
+
+ // If device opens for both playback and capture, we determine the channels.
+ if ( info.outputChannels > 0 && info.inputChannels > 0 )
+ info.duplexChannels = (info.outputChannels > info.inputChannels) ? info.inputChannels : info.outputChannels;
+
+ if ( device == 0 ) info.isDefaultInput = true;
+
+ // Copy name and return.
+ info.name = dsDevices[ device ].name;
+ info.probed = true;
+ return info;
+}
+
+bool RtApiDs :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,
+ unsigned int firstChannel, unsigned int sampleRate,
+ RtAudioFormat format, unsigned int *bufferSize,
+ RtAudio::StreamOptions *options )
+{
+ if ( channels + firstChannel > 2 ) {
+ errorText_ = "RtApiDs::probeDeviceOpen: DirectSound does not support more than 2 channels per device.";
+ return FAILURE;
+ }
+
+ unsigned int nDevices = dsDevices.size();
+ if ( nDevices == 0 ) {
+ // This should not happen because a check is made before this function is called.
+ errorText_ = "RtApiDs::probeDeviceOpen: no devices found!";
+ return FAILURE;
+ }
+
+ if ( device >= nDevices ) {
+ // This should not happen because a check is made before this function is called.
+ errorText_ = "RtApiDs::probeDeviceOpen: device ID is invalid!";
+ return FAILURE;
+ }
+
+ if ( mode == OUTPUT ) {
+ if ( dsDevices[ device ].validId[0] == false ) {
+ errorStream_ << "RtApiDs::probeDeviceOpen: device (" << device << ") does not support output!";
errorText_ = errorStream_.str();
- error( RtError::WARNING );
+ return FAILURE;
+ }
+ }
+ else { // mode == INPUT
+ if ( dsDevices[ device ].validId[1] == false ) {
+ errorStream_ << "RtApiDs::probeDeviceOpen: device (" << device << ") does not support input!";
+ errorText_ = errorStream_.str();
+ return FAILURE;
}
+ }
- if ( dsinfo.name.empty() ) goto probeInput;
+ // According to a note in PortAudio, using GetDesktopWindow()
+ // instead of GetForegroundWindow() is supposed to avoid problems
+ // that occur when the application's window is not the foreground
+ // window. Also, if the application window closes before the
+ // DirectSound buffer, DirectSound can crash. In the past, I had
+ // problems when using GetDesktopWindow() but it seems fine now
+ // (January 2010). I'll leave it commented here.
+ // HWND hWnd = GetForegroundWindow();
+ HWND hWnd = GetDesktopWindow();
+
+ // Check the numberOfBuffers parameter and limit the lowest value to
+ // two. This is a judgement call and a value of two is probably too
+ // low for capture, but it should work for playback.
+ int nBuffers = 0;
+ if ( options ) nBuffers = options->numberOfBuffers;
+ if ( options && options->flags & RTAUDIO_MINIMIZE_LATENCY ) nBuffers = 2;
+ if ( nBuffers < 2 ) nBuffers = 3;
+
+ // Check the lower range of the user-specified buffer size and set
+ // (arbitrarily) to a lower bound of 32.
+ if ( *bufferSize < 32 ) *bufferSize = 32;
+
+ // Create the wave format structure. The data format setting will
+ // be determined later.
+ WAVEFORMATEX waveFormat;
+ ZeroMemory( &waveFormat, sizeof(WAVEFORMATEX) );
+ waveFormat.wFormatTag = WAVE_FORMAT_PCM;
+ waveFormat.nChannels = channels + firstChannel;
+ waveFormat.nSamplesPerSec = (unsigned long) sampleRate;
+
+ // Determine the device buffer size. By default, we'll use the value
+ // defined above (32K), but we will grow it to make allowances for
+ // very large software buffer sizes.
+ DWORD dsBufferSize = MINIMUM_DEVICE_BUFFER_SIZE;;
+ DWORD dsPointerLeadTime = 0;
+
+ void *ohandle = 0, *bhandle = 0;
+ HRESULT result;
+ if ( mode == OUTPUT ) {
LPDIRECTSOUND output;
- DSCAPS outCaps;
- result = DirectSoundCreate( dsinfo.id, &output, NULL );
+ result = DirectSoundCreate( dsDevices[ device ].id[0], &output, NULL );
if ( FAILED( result ) ) {
- errorStream_ << "RtApiDs::getDeviceInfo: error (" << getErrorString( result ) << ") opening output device (" << dsinfo.name << ")!";
+ errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") opening output device (" << dsDevices[ device ].name << ")!";
errorText_ = errorStream_.str();
- error( RtError::WARNING );
- return info;
+ return FAILURE;
}
+ DSCAPS outCaps;
outCaps.dwSize = sizeof( outCaps );
result = output->GetCaps( &outCaps );
if ( FAILED( result ) ) {
output->Release();
- errorStream_ << "RtApiDs::getDeviceInfo: error (" << getErrorString( result ) << ") getting capabilities!";
+ errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") getting capabilities (" << dsDevices[ device ].name << ")!";
errorText_ = errorStream_.str();
- error( RtError::WARNING );
- return info;
+ return FAILURE;
}
- // Get output channel information.
- info.outputChannels = ( outCaps.dwFlags & DSCAPS_PRIMARYSTEREO ) ? 2 : 1;
-
- // Get sample rate information.
- info.sampleRates.clear();
- for ( unsigned int k=0; k<MAX_SAMPLE_RATES; k++ ) {
- if ( SAMPLE_RATES[k] >= (unsigned int) outCaps.dwMinSecondarySampleRate &&
- SAMPLE_RATES[k] <= (unsigned int) outCaps.dwMaxSecondarySampleRate )
- info.sampleRates.push_back( SAMPLE_RATES[k] );
+ // Check channel information.
+ if ( channels + firstChannel == 2 && !( outCaps.dwFlags & DSCAPS_PRIMARYSTEREO ) ) {
+ errorStream_ << "RtApiDs::getDeviceInfo: the output device (" << dsDevices[ device ].name << ") does not support stereo playback.";
+ errorText_ = errorStream_.str();
+ return FAILURE;
}
- // Get format information.
- if ( outCaps.dwFlags & DSCAPS_PRIMARY16BIT ) info.nativeFormats |= RTAUDIO_SINT16;
- if ( outCaps.dwFlags & DSCAPS_PRIMARY8BIT ) info.nativeFormats |= RTAUDIO_SINT8;
-
- output->Release();
-
- if ( getDefaultOutputDevice() == device )
- info.isDefaultOutput = true;
-
- // Copy name and return.
- info.name = dsinfo.name;
+ // Check format information. Use 16-bit format unless not
+ // supported or user requests 8-bit.
+ if ( outCaps.dwFlags & DSCAPS_PRIMARY16BIT &&
+ !( format == RTAUDIO_SINT8 && outCaps.dwFlags & DSCAPS_PRIMARY8BIT ) ) {
+ waveFormat.wBitsPerSample = 16;
+ stream_.deviceFormat[mode] = RTAUDIO_SINT16;
+ }
+ else {
+ waveFormat.wBitsPerSample = 8;
+ stream_.deviceFormat[mode] = RTAUDIO_SINT8;
+ }
+ stream_.userFormat = format;
- info.probed = true;
- return info;
+ // Update wave format structure and buffer information.
+ waveFormat.nBlockAlign = waveFormat.nChannels * waveFormat.wBitsPerSample / 8;
+ waveFormat.nAvgBytesPerSec = waveFormat.nSamplesPerSec * waveFormat.nBlockAlign;
+ dsPointerLeadTime = nBuffers * (*bufferSize) * (waveFormat.wBitsPerSample / 8) * channels;
- probeInput:
+ // If the user wants an even bigger buffer, increase the device buffer size accordingly.
+ while ( dsPointerLeadTime * 2U > dsBufferSize )
+ dsBufferSize *= 2;
- dsinfo.isInput = true;
- result = DirectSoundCaptureEnumerate( (LPDSENUMCALLBACK) deviceQueryCallback, &dsinfo );
+ // Set cooperative level to DSSCL_EXCLUSIVE ... sound stops when window focus changes.
+ // result = output->SetCooperativeLevel( hWnd, DSSCL_EXCLUSIVE );
+ // Set cooperative level to DSSCL_PRIORITY ... sound remains when window focus changes.
+ result = output->SetCooperativeLevel( hWnd, DSSCL_PRIORITY );
if ( FAILED( result ) ) {
- errorStream_ << "RtApiDs::getDeviceInfo: error (" << getErrorString( result ) << ") enumerating input devices!";
+ output->Release();
+ errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") setting cooperative level (" << dsDevices[ device ].name << ")!";
errorText_ = errorStream_.str();
- error( RtError::WARNING );
+ return FAILURE;
}
- if ( dsinfo.name.empty() ) return info;
+ // Even though we will write to the secondary buffer, we need to
+ // access the primary buffer to set the correct output format
+ // (since the default is 8-bit, 22 kHz!). Setup the DS primary
+ // buffer description.
+ DSBUFFERDESC bufferDescription;
+ ZeroMemory( &bufferDescription, sizeof( DSBUFFERDESC ) );
+ bufferDescription.dwSize = sizeof( DSBUFFERDESC );
+ bufferDescription.dwFlags = DSBCAPS_PRIMARYBUFFER;
- LPDIRECTSOUNDCAPTURE input;
- result = DirectSoundCaptureCreate( dsinfo.id, &input, NULL );
+ // Obtain the primary buffer
+ LPDIRECTSOUNDBUFFER buffer;
+ result = output->CreateSoundBuffer( &bufferDescription, &buffer, NULL );
if ( FAILED( result ) ) {
- errorStream_ << "RtApiDs::getDeviceInfo: error (" << getErrorString( result ) << ") opening input device (" << dsinfo.name << ")!";
+ output->Release();
+ errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") accessing primary buffer (" << dsDevices[ device ].name << ")!";
errorText_ = errorStream_.str();
- error( RtError::WARNING );
- return info;
+ return FAILURE;
}
- DSCCAPS inCaps;
- inCaps.dwSize = sizeof( inCaps );
- result = input->GetCaps( &inCaps );
+ // Set the primary DS buffer sound format.
+ result = buffer->SetFormat( &waveFormat );
if ( FAILED( result ) ) {
- input->Release();
- errorStream_ << "RtApiDs::getDeviceInfo: error (" << getErrorString( result ) << ") getting object capabilities (" << dsinfo.name << ")!";
+ output->Release();
+ errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") setting primary buffer format (" << dsDevices[ device ].name << ")!";
errorText_ = errorStream_.str();
- error( RtError::WARNING );
- return info;
+ return FAILURE;
}
- // Get input channel information.
- info.inputChannels = inCaps.dwChannels;
-
- // Get sample rate and format information.
- if ( inCaps.dwChannels == 2 ) {
- if ( inCaps.dwFormats & WAVE_FORMAT_1S16 ) info.nativeFormats |= RTAUDIO_SINT16;
- if ( inCaps.dwFormats & WAVE_FORMAT_2S16 ) info.nativeFormats |= RTAUDIO_SINT16;
- if ( inCaps.dwFormats & WAVE_FORMAT_4S16 ) info.nativeFormats |= RTAUDIO_SINT16;
- if ( inCaps.dwFormats & WAVE_FORMAT_96S16 ) info.nativeFormats |= RTAUDIO_SINT16;
- if ( inCaps.dwFormats & WAVE_FORMAT_1S08 ) info.nativeFormats |= RTAUDIO_SINT8;
- if ( inCaps.dwFormats & WAVE_FORMAT_2S08 ) info.nativeFormats |= RTAUDIO_SINT8;
- if ( inCaps.dwFormats & WAVE_FORMAT_4S08 ) info.nativeFormats |= RTAUDIO_SINT8;
- if ( inCaps.dwFormats & WAVE_FORMAT_96S08 ) info.nativeFormats |= RTAUDIO_SINT8;
-
- if ( info.nativeFormats & RTAUDIO_SINT16 ) {
- if ( inCaps.dwFormats & WAVE_FORMAT_1S16 ) info.sampleRates.push_back( 11025 );
- if ( inCaps.dwFormats & WAVE_FORMAT_2S16 ) info.sampleRates.push_back( 22050 );
- if ( inCaps.dwFormats & WAVE_FORMAT_4S16 ) info.sampleRates.push_back( 44100 );
- if ( inCaps.dwFormats & WAVE_FORMAT_96S16 ) info.sampleRates.push_back( 96000 );
- }
- else if ( info.nativeFormats & RTAUDIO_SINT8 ) {
- if ( inCaps.dwFormats & WAVE_FORMAT_1S08 ) info.sampleRates.push_back( 11025 );
- if ( inCaps.dwFormats & WAVE_FORMAT_2S08 ) info.sampleRates.push_back( 22050 );
- if ( inCaps.dwFormats & WAVE_FORMAT_4S08 ) info.sampleRates.push_back( 44100 );
- if ( inCaps.dwFormats & WAVE_FORMAT_96S08 ) info.sampleRates.push_back( 44100 );
+ // Setup the secondary DS buffer description.
+ ZeroMemory( &bufferDescription, sizeof( DSBUFFERDESC ) );
+ bufferDescription.dwSize = sizeof( DSBUFFERDESC );
+ bufferDescription.dwFlags = ( DSBCAPS_STICKYFOCUS |
+ DSBCAPS_GLOBALFOCUS |
+ DSBCAPS_GETCURRENTPOSITION2 |
+ DSBCAPS_LOCHARDWARE ); // Force hardware mixing
+ bufferDescription.dwBufferBytes = dsBufferSize;
+ bufferDescription.lpwfxFormat = &waveFormat;
+
+ // Try to create the secondary DS buffer. If that doesn't work,
+ // try to use software mixing. Otherwise, there's a problem.
+ result = output->CreateSoundBuffer( &bufferDescription, &buffer, NULL );
+ if ( FAILED( result ) ) {
+ bufferDescription.dwFlags = ( DSBCAPS_STICKYFOCUS |
+ DSBCAPS_GLOBALFOCUS |
+ DSBCAPS_GETCURRENTPOSITION2 |
+ DSBCAPS_LOCSOFTWARE ); // Force software mixing
+ result = output->CreateSoundBuffer( &bufferDescription, &buffer, NULL );
+ if ( FAILED( result ) ) {
+ output->Release();
+ errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") creating secondary buffer (" << dsDevices[ device ].name << ")!";
+ errorText_ = errorStream_.str();
+ return FAILURE;
}
}
- else if ( inCaps.dwChannels == 1 ) {
- if ( inCaps.dwFormats & WAVE_FORMAT_1M16 ) info.nativeFormats |= RTAUDIO_SINT16;
- if ( inCaps.dwFormats & WAVE_FORMAT_2M16 ) info.nativeFormats |= RTAUDIO_SINT16;
- if ( inCaps.dwFormats & WAVE_FORMAT_4M16 ) info.nativeFormats |= RTAUDIO_SINT16;
- if ( inCaps.dwFormats & WAVE_FORMAT_96M16 ) info.nativeFormats |= RTAUDIO_SINT16;
- if ( inCaps.dwFormats & WAVE_FORMAT_1M08 ) info.nativeFormats |= RTAUDIO_SINT8;
- if ( inCaps.dwFormats & WAVE_FORMAT_2M08 ) info.nativeFormats |= RTAUDIO_SINT8;
- if ( inCaps.dwFormats & WAVE_FORMAT_4M08 ) info.nativeFormats |= RTAUDIO_SINT8;
- if ( inCaps.dwFormats & WAVE_FORMAT_96M08 ) info.nativeFormats |= RTAUDIO_SINT8;
-
- if ( info.nativeFormats & RTAUDIO_SINT16 ) {
- if ( inCaps.dwFormats & WAVE_FORMAT_1M16 ) info.sampleRates.push_back( 11025 );
- if ( inCaps.dwFormats & WAVE_FORMAT_2M16 ) info.sampleRates.push_back( 22050 );
- if ( inCaps.dwFormats & WAVE_FORMAT_4M16 ) info.sampleRates.push_back( 44100 );
- if ( inCaps.dwFormats & WAVE_FORMAT_96M16 ) info.sampleRates.push_back( 96000 );
- }
- else if ( info.nativeFormats & RTAUDIO_SINT8 ) {
- if ( inCaps.dwFormats & WAVE_FORMAT_1M08 ) info.sampleRates.push_back( 11025 );
- if ( inCaps.dwFormats & WAVE_FORMAT_2M08 ) info.sampleRates.push_back( 22050 );
- if ( inCaps.dwFormats & WAVE_FORMAT_4M08 ) info.sampleRates.push_back( 44100 );
- if ( inCaps.dwFormats & WAVE_FORMAT_96M08 ) info.sampleRates.push_back( 96000 );
- }
+
+ // Get the buffer size ... might be different from what we specified.
+ DSBCAPS dsbcaps;
+ dsbcaps.dwSize = sizeof( DSBCAPS );
+ result = buffer->GetCaps( &dsbcaps );
+ if ( FAILED( result ) ) {
+ output->Release();
+ buffer->Release();
+ errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") getting buffer settings (" << dsDevices[ device ].name << ")!";
+ errorText_ = errorStream_.str();
+ return FAILURE;
}
- else info.inputChannels = 0; // technically, this would be an error
- input->Release();
+ dsBufferSize = dsbcaps.dwBufferBytes;
- if ( info.inputChannels == 0 ) return info;
+ // Lock the DS buffer
+ LPVOID audioPtr;
+ DWORD dataLen;
+ result = buffer->Lock( 0, dsBufferSize, &audioPtr, &dataLen, NULL, NULL, 0 );
+ if ( FAILED( result ) ) {
+ output->Release();
+ buffer->Release();
+ errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") locking buffer (" << dsDevices[ device ].name << ")!";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
- if ( getDefaultInputDevice() == device )
- info.isDefaultInput = true;
+ // Zero the DS buffer
+ ZeroMemory( audioPtr, dataLen );
- // Copy name and return.
- info.name = dsinfo.name;
- info.probed = true;
- return info;
+ // Unlock the DS buffer
+ result = buffer->Unlock( audioPtr, dataLen, NULL, 0 );
+ if ( FAILED( result ) ) {
+ output->Release();
+ buffer->Release();
+ errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") unlocking buffer (" << dsDevices[ device ].name << ")!";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
+
+ ohandle = (void *) output;
+ bhandle = (void *) buffer;
}
- bool RtApiDs :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,
- unsigned int firstChannel, unsigned int sampleRate,
- RtAudioFormat format, unsigned int *bufferSize,
- RtAudio::StreamOptions *options )
- {
- if ( channels + firstChannel > 2 ) {
- errorText_ = "RtApiDs::probeDeviceOpen: DirectSound does not support more than 2 channels per device.";
+ if ( mode == INPUT ) {
+
+ LPDIRECTSOUNDCAPTURE input;
+ result = DirectSoundCaptureCreate( dsDevices[ device ].id[1], &input, NULL );
+ if ( FAILED( result ) ) {
+ errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") opening input device (" << dsDevices[ device ].name << ")!";
+ errorText_ = errorStream_.str();
return FAILURE;
}
- // Enumerate through devices to find the id (if it exists). Note
- // that we have to do the output enumeration first, even if this is
- // an input device, in order for the device counter to be correct.
- EnumInfo dsinfo;
- dsinfo.findIndex = true;
- dsinfo.index = device;
- HRESULT result = DirectSoundEnumerate( (LPDSENUMCALLBACK) deviceQueryCallback, &dsinfo );
+ DSCCAPS inCaps;
+ inCaps.dwSize = sizeof( inCaps );
+ result = input->GetCaps( &inCaps );
if ( FAILED( result ) ) {
- errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") enumerating output devices!";
+ input->Release();
+ errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") getting input capabilities (" << dsDevices[ device ].name << ")!";
errorText_ = errorStream_.str();
return FAILURE;
}
- if ( mode == OUTPUT ) {
- if ( dsinfo.name.empty() ) {
- errorStream_ << "RtApiDs::probeDeviceOpen: device (" << device << ") does not support output!";
- errorText_ = errorStream_.str();
- return FAILURE;
- }
- }
- else { // mode == INPUT
- dsinfo.isInput = true;
- HRESULT result = DirectSoundCaptureEnumerate( (LPDSENUMCALLBACK) deviceQueryCallback, &dsinfo );
- if ( FAILED( result ) ) {
- errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") enumerating input devices!";
- errorText_ = errorStream_.str();
- return FAILURE;
- }
- if ( dsinfo.name.empty() ) {
- errorStream_ << "RtApiDs::probeDeviceOpen: device (" << device << ") does not support input!";
- errorText_ = errorStream_.str();
- return FAILURE;
- }
+ // Check channel information.
+ if ( inCaps.dwChannels < channels + firstChannel ) {
+ errorText_ = "RtApiDs::getDeviceInfo: the input device does not support requested input channels.";
+ return FAILURE;
}
- // According to a note in PortAudio, using GetDesktopWindow()
- // instead of GetForegroundWindow() is supposed to avoid problems
- // that occur when the application's window is not the foreground
- // window. Also, if the application window closes before the
- // DirectSound buffer, DirectSound can crash. However, for console
- // applications, no sound was produced when using GetDesktopWindow().
- HWND hWnd = GetForegroundWindow();
-
- // Check the numberOfBuffers parameter and limit the lowest value to
- // two. This is a judgement call and a value of two is probably too
- // low for capture, but it should work for playback.
- int nBuffers = 0;
- if ( options ) nBuffers = options->numberOfBuffers;
- if ( options && options->flags & RTAUDIO_MINIMIZE_LATENCY ) nBuffers = 2;
- if ( nBuffers < 2 ) nBuffers = 3;
-
- // Create the wave format structure. The data format setting will
- // be determined later.
- WAVEFORMATEX waveFormat;
- ZeroMemory( &waveFormat, sizeof(WAVEFORMATEX) );
- waveFormat.wFormatTag = WAVE_FORMAT_PCM;
- waveFormat.nChannels = channels + firstChannel;
- waveFormat.nSamplesPerSec = (unsigned long) sampleRate;
-
- // Determine the device buffer size. By default, 32k, but we will
- // grow it to make allowances for very large software buffer sizes.
- DWORD dsBufferSize = 0;
- DWORD dsPointerLeadTime = 0;
- long bufferBytes = MINIMUM_DEVICE_BUFFER_SIZE; // sound cards will always *knock wood* support this
-
- void *ohandle = 0, *bhandle = 0;
- if ( mode == OUTPUT ) {
-
- LPDIRECTSOUND output;
- result = DirectSoundCreate( dsinfo.id, &output, NULL );
- if ( FAILED( result ) ) {
- errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") opening output device (" << dsinfo.name << ")!";
- errorText_ = errorStream_.str();
- return FAILURE;
- }
-
- DSCAPS outCaps;
- outCaps.dwSize = sizeof( outCaps );
- result = output->GetCaps( &outCaps );
- if ( FAILED( result ) ) {
- output->Release();
- errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") getting capabilities (" << dsinfo.name << ")!";
- errorText_ = errorStream_.str();
- return FAILURE;
- }
-
- // Check channel information.
- if ( channels + firstChannel == 2 && !( outCaps.dwFlags & DSCAPS_PRIMARYSTEREO ) ) {
- errorStream_ << "RtApiDs::getDeviceInfo: the output device (" << dsinfo.name << ") does not support stereo playback.";
- errorText_ = errorStream_.str();
- return FAILURE;
+ // Check format information. Use 16-bit format unless user
+ // requests 8-bit.
+ DWORD deviceFormats;
+ if ( channels + firstChannel == 2 ) {
+ deviceFormats = WAVE_FORMAT_1S08 | WAVE_FORMAT_2S08 | WAVE_FORMAT_4S08 | WAVE_FORMAT_96S08;
+ if ( format == RTAUDIO_SINT8 && inCaps.dwFormats & deviceFormats ) {
+ waveFormat.wBitsPerSample = 8;
+ stream_.deviceFormat[mode] = RTAUDIO_SINT8;
}
-
- // Check format information. Use 16-bit format unless not
- // supported or user requests 8-bit.
- if ( outCaps.dwFlags & DSCAPS_PRIMARY16BIT &&
- !( format == RTAUDIO_SINT8 && outCaps.dwFlags & DSCAPS_PRIMARY8BIT ) ) {
+ else { // assume 16-bit is supported
waveFormat.wBitsPerSample = 16;
stream_.deviceFormat[mode] = RTAUDIO_SINT16;
}
- else {
+ }
+ else { // channel == 1
+ deviceFormats = WAVE_FORMAT_1M08 | WAVE_FORMAT_2M08 | WAVE_FORMAT_4M08 | WAVE_FORMAT_96M08;
+ if ( format == RTAUDIO_SINT8 && inCaps.dwFormats & deviceFormats ) {
waveFormat.wBitsPerSample = 8;
stream_.deviceFormat[mode] = RTAUDIO_SINT8;
}
- stream_.userFormat = format;
-
- // Update wave format structure and buffer information.
- waveFormat.nBlockAlign = waveFormat.nChannels * waveFormat.wBitsPerSample / 8;
- waveFormat.nAvgBytesPerSec = waveFormat.nSamplesPerSec * waveFormat.nBlockAlign;
- dsPointerLeadTime = nBuffers * (*bufferSize) * (waveFormat.wBitsPerSample / 8) * channels;
-
- // If the user wants an even bigger buffer, increase the device buffer size accordingly.
- while ( dsPointerLeadTime * 2U > (DWORD) bufferBytes )
- bufferBytes *= 2;
-
- // Set cooperative level to DSSCL_EXCLUSIVE ... sound stops when window focus changes.
- //result = output->SetCooperativeLevel( hWnd, DSSCL_EXCLUSIVE );
- // Set cooperative level to DSSCL_PRIORITY ... sound remains when window focus changes.
- result = output->SetCooperativeLevel( hWnd, DSSCL_PRIORITY );
- if ( FAILED( result ) ) {
- output->Release();
- errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") setting cooperative level (" << dsinfo.name << ")!";
- errorText_ = errorStream_.str();
- return FAILURE;
- }
-
- // Even though we will write to the secondary buffer, we need to
- // access the primary buffer to set the correct output format
- // (since the default is 8-bit, 22 kHz!). Setup the DS primary
- // buffer description.
- DSBUFFERDESC bufferDescription;
- ZeroMemory( &bufferDescription, sizeof( DSBUFFERDESC ) );
- bufferDescription.dwSize = sizeof( DSBUFFERDESC );
- bufferDescription.dwFlags = DSBCAPS_PRIMARYBUFFER;
-
- // Obtain the primary buffer
- LPDIRECTSOUNDBUFFER buffer;
- result = output->CreateSoundBuffer( &bufferDescription, &buffer, NULL );
- if ( FAILED( result ) ) {
- output->Release();
- errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") accessing primary buffer (" << dsinfo.name << ")!";
- errorText_ = errorStream_.str();
- return FAILURE;
- }
-
- // Set the primary DS buffer sound format.
- result = buffer->SetFormat( &waveFormat );
- if ( FAILED( result ) ) {
- output->Release();
- errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") setting primary buffer format (" << dsinfo.name << ")!";
- errorText_ = errorStream_.str();
- return FAILURE;
+ else { // assume 16-bit is supported
+ waveFormat.wBitsPerSample = 16;
+ stream_.deviceFormat[mode] = RTAUDIO_SINT16;
}
+ }
+ stream_.userFormat = format;
- // Setup the secondary DS buffer description.
- dsBufferSize = (DWORD) bufferBytes;
- ZeroMemory( &bufferDescription, sizeof( DSBUFFERDESC ) );
- bufferDescription.dwSize = sizeof( DSBUFFERDESC );
- bufferDescription.dwFlags = ( DSBCAPS_STICKYFOCUS |
- DSBCAPS_GLOBALFOCUS |
- DSBCAPS_GETCURRENTPOSITION2 |
- DSBCAPS_LOCHARDWARE ); // Force hardware mixing
- bufferDescription.dwBufferBytes = bufferBytes;
- bufferDescription.lpwfxFormat = &waveFormat;
-
- // Try to create the secondary DS buffer. If that doesn't work,
- // try to use software mixing. Otherwise, there's a problem.
- result = output->CreateSoundBuffer( &bufferDescription, &buffer, NULL );
- if ( FAILED( result ) ) {
- bufferDescription.dwFlags = ( DSBCAPS_STICKYFOCUS |
- DSBCAPS_GLOBALFOCUS |
- DSBCAPS_GETCURRENTPOSITION2 |
- DSBCAPS_LOCSOFTWARE ); // Force software mixing
- result = output->CreateSoundBuffer( &bufferDescription, &buffer, NULL );
- if ( FAILED( result ) ) {
- output->Release();
- errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") creating secondary buffer (" << dsinfo.name << ")!";
- errorText_ = errorStream_.str();
- return FAILURE;
- }
- }
+ // Update wave format structure and buffer information.
+ waveFormat.nBlockAlign = waveFormat.nChannels * waveFormat.wBitsPerSample / 8;
+ waveFormat.nAvgBytesPerSec = waveFormat.nSamplesPerSec * waveFormat.nBlockAlign;
+ dsPointerLeadTime = nBuffers * (*bufferSize) * (waveFormat.wBitsPerSample / 8) * channels;
+
+ // If the user wants an even bigger buffer, increase the device buffer size accordingly.
+ while ( dsPointerLeadTime * 2U > dsBufferSize )
+ dsBufferSize *= 2;
+
+ // Setup the secondary DS buffer description.
+ DSCBUFFERDESC bufferDescription;
+ ZeroMemory( &bufferDescription, sizeof( DSCBUFFERDESC ) );
+ bufferDescription.dwSize = sizeof( DSCBUFFERDESC );
+ bufferDescription.dwFlags = 0;
+ bufferDescription.dwReserved = 0;
+ bufferDescription.dwBufferBytes = dsBufferSize;
+ bufferDescription.lpwfxFormat = &waveFormat;
+
+ // Create the capture buffer.
+ LPDIRECTSOUNDCAPTUREBUFFER buffer;
+ result = input->CreateCaptureBuffer( &bufferDescription, &buffer, NULL );
+ if ( FAILED( result ) ) {
+ input->Release();
+ errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") creating input buffer (" << dsDevices[ device ].name << ")!";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
- // Get the buffer size ... might be different from what we specified.
- DSBCAPS dsbcaps;
- dsbcaps.dwSize = sizeof( DSBCAPS );
- result = buffer->GetCaps( &dsbcaps );
- if ( FAILED( result ) ) {
- output->Release();
- buffer->Release();
- errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") getting buffer settings (" << dsinfo.name << ")!";
- errorText_ = errorStream_.str();
- return FAILURE;
- }
+ // Get the buffer size ... might be different from what we specified.
+ DSCBCAPS dscbcaps;
+ dscbcaps.dwSize = sizeof( DSCBCAPS );
+ result = buffer->GetCaps( &dscbcaps );
+ if ( FAILED( result ) ) {
+ input->Release();
+ buffer->Release();
+ errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") getting buffer settings (" << dsDevices[ device ].name << ")!";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
- bufferBytes = dsbcaps.dwBufferBytes;
+ dsBufferSize = dscbcaps.dwBufferBytes;
- // Lock the DS buffer
- LPVOID audioPtr;
- DWORD dataLen;
- result = buffer->Lock( 0, bufferBytes, &audioPtr, &dataLen, NULL, NULL, 0 );
- if ( FAILED( result ) ) {
- output->Release();
- buffer->Release();
- errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") locking buffer (" << dsinfo.name << ")!";
- errorText_ = errorStream_.str();
- return FAILURE;
- }
+ // NOTE: We could have a problem here if this is a duplex stream
+ // and the play and capture hardware buffer sizes are different
+ // (I'm actually not sure if that is a problem or not).
+ // Currently, we are not verifying that.
- // Zero the DS buffer
- ZeroMemory( audioPtr, dataLen );
+ // Lock the capture buffer
+ LPVOID audioPtr;
+ DWORD dataLen;
+ result = buffer->Lock( 0, dsBufferSize, &audioPtr, &dataLen, NULL, NULL, 0 );
+ if ( FAILED( result ) ) {
+ input->Release();
+ buffer->Release();
+ errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") locking input buffer (" << dsDevices[ device ].name << ")!";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
- // Unlock the DS buffer
- result = buffer->Unlock( audioPtr, dataLen, NULL, 0 );
- if ( FAILED( result ) ) {
- output->Release();
- buffer->Release();
- errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") unlocking buffer (" << dsinfo.name << ")!";
- errorText_ = errorStream_.str();
- return FAILURE;
- }
+ // Zero the buffer
+ ZeroMemory( audioPtr, dataLen );
- dsBufferSize = bufferBytes;
- ohandle = (void *) output;
- bhandle = (void *) buffer;
+ // Unlock the buffer
+ result = buffer->Unlock( audioPtr, dataLen, NULL, 0 );
+ if ( FAILED( result ) ) {
+ input->Release();
+ buffer->Release();
+ errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") unlocking input buffer (" << dsDevices[ device ].name << ")!";
+ errorText_ = errorStream_.str();
+ return FAILURE;
}
- if ( mode == INPUT ) {
+ ohandle = (void *) input;
+ bhandle = (void *) buffer;
+ }
- LPDIRECTSOUNDCAPTURE input;
- result = DirectSoundCaptureCreate( dsinfo.id, &input, NULL );
- if ( FAILED( result ) ) {
- errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") opening input device (" << dsinfo.name << ")!";
- errorText_ = errorStream_.str();
- return FAILURE;
- }
+ // Set various stream parameters
+ DsHandle *handle = 0;
+ stream_.nDeviceChannels[mode] = channels + firstChannel;
+ stream_.nUserChannels[mode] = channels;
+ stream_.bufferSize = *bufferSize;
+ stream_.channelOffset[mode] = firstChannel;
+ stream_.deviceInterleaved[mode] = true;
+ if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) stream_.userInterleaved = false;
+ else stream_.userInterleaved = true;
- DSCCAPS inCaps;
- inCaps.dwSize = sizeof( inCaps );
- result = input->GetCaps( &inCaps );
- if ( FAILED( result ) ) {
- input->Release();
- errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") getting input capabilities (" << dsinfo.name << ")!";
- errorText_ = errorStream_.str();
- return FAILURE;
- }
+ // Set flag for buffer conversion
+ stream_.doConvertBuffer[mode] = false;
+ if (stream_.nUserChannels[mode] != stream_.nDeviceChannels[mode])
+ stream_.doConvertBuffer[mode] = true;
+ if (stream_.userFormat != stream_.deviceFormat[mode])
+ stream_.doConvertBuffer[mode] = true;
+ if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] &&
+ stream_.nUserChannels[mode] > 1 )
+ stream_.doConvertBuffer[mode] = true;
- // Check channel information.
- if ( inCaps.dwChannels < channels + firstChannel ) {
- errorText_ = "RtApiDs::getDeviceInfo: the input device does not support requested input channels.";
- return FAILURE;
- }
+ // Allocate necessary internal buffers
+ long bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat );
+ stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 );
+ if ( stream_.userBuffer[mode] == NULL ) {
+ errorText_ = "RtApiDs::probeDeviceOpen: error allocating user buffer memory.";
+ goto error;
+ }
- // Check format information. Use 16-bit format unless user
- // requests 8-bit.
- DWORD deviceFormats;
- if ( channels + firstChannel == 2 ) {
- deviceFormats = WAVE_FORMAT_1S08 | WAVE_FORMAT_2S08 | WAVE_FORMAT_4S08 | WAVE_FORMAT_96S08;
- if ( format == RTAUDIO_SINT8 && inCaps.dwFormats & deviceFormats ) {
- waveFormat.wBitsPerSample = 8;
- stream_.deviceFormat[mode] = RTAUDIO_SINT8;
- }
- else { // assume 16-bit is supported
- waveFormat.wBitsPerSample = 16;
- stream_.deviceFormat[mode] = RTAUDIO_SINT16;
- }
- }
- else { // channel == 1
- deviceFormats = WAVE_FORMAT_1M08 | WAVE_FORMAT_2M08 | WAVE_FORMAT_4M08 | WAVE_FORMAT_96M08;
- if ( format == RTAUDIO_SINT8 && inCaps.dwFormats & deviceFormats ) {
- waveFormat.wBitsPerSample = 8;
- stream_.deviceFormat[mode] = RTAUDIO_SINT8;
- }
- else { // assume 16-bit is supported
- waveFormat.wBitsPerSample = 16;
- stream_.deviceFormat[mode] = RTAUDIO_SINT16;
- }
- }
- stream_.userFormat = format;
-
- // Update wave format structure and buffer information.
- waveFormat.nBlockAlign = waveFormat.nChannels * waveFormat.wBitsPerSample / 8;
- waveFormat.nAvgBytesPerSec = waveFormat.nSamplesPerSec * waveFormat.nBlockAlign;
-
- // Setup the secondary DS buffer description.
- dsBufferSize = bufferBytes;
- DSCBUFFERDESC bufferDescription;
- ZeroMemory( &bufferDescription, sizeof( DSCBUFFERDESC ) );
- bufferDescription.dwSize = sizeof( DSCBUFFERDESC );
- bufferDescription.dwFlags = 0;
- bufferDescription.dwReserved = 0;
- bufferDescription.dwBufferBytes = bufferBytes;
- bufferDescription.lpwfxFormat = &waveFormat;
-
- // Create the capture buffer.
- LPDIRECTSOUNDCAPTUREBUFFER buffer;
- result = input->CreateCaptureBuffer( &bufferDescription, &buffer, NULL );
- if ( FAILED( result ) ) {
- input->Release();
- errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") creating input buffer (" << dsinfo.name << ")!";
- errorText_ = errorStream_.str();
- return FAILURE;
- }
+ if ( stream_.doConvertBuffer[mode] ) {
- // Lock the capture buffer
- LPVOID audioPtr;
- DWORD dataLen;
- result = buffer->Lock( 0, bufferBytes, &audioPtr, &dataLen, NULL, NULL, 0 );
- if ( FAILED( result ) ) {
- input->Release();
- buffer->Release();
- errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") locking input buffer (" << dsinfo.name << ")!";
- errorText_ = errorStream_.str();
- return FAILURE;
+ bool makeBuffer = true;
+ bufferBytes = stream_.nDeviceChannels[mode] * formatBytes( stream_.deviceFormat[mode] );
+ if ( mode == INPUT ) {
+ if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) {
+ unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] );
+ if ( bufferBytes <= (long) bytesOut ) makeBuffer = false;
}
+ }
- // Zero the buffer
- ZeroMemory( audioPtr, dataLen );
-
- // Unlock the buffer
- result = buffer->Unlock( audioPtr, dataLen, NULL, 0 );
- if ( FAILED( result ) ) {
- input->Release();
- buffer->Release();
- errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") unlocking input buffer (" << dsinfo.name << ")!";
- errorText_ = errorStream_.str();
- return FAILURE;
+ if ( makeBuffer ) {
+ bufferBytes *= *bufferSize;
+ if ( stream_.deviceBuffer ) free( stream_.deviceBuffer );
+ stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 );
+ if ( stream_.deviceBuffer == NULL ) {
+ errorText_ = "RtApiDs::probeDeviceOpen: error allocating device buffer memory.";
+ goto error;
}
-
- dsBufferSize = bufferBytes;
- ohandle = (void *) input;
- bhandle = (void *) buffer;
}
+ }
- // Set various stream parameters
- DsHandle *handle = 0;
- stream_.nDeviceChannels[mode] = channels + firstChannel;
- stream_.nUserChannels[mode] = channels;
- stream_.bufferSize = *bufferSize;
- stream_.channelOffset[mode] = firstChannel;
- stream_.deviceInterleaved[mode] = true;
- if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) stream_.userInterleaved = false;
- else stream_.userInterleaved = true;
-
- // Set flag for buffer conversion
- stream_.doConvertBuffer[mode] = false;
- if (stream_.nUserChannels[mode] != stream_.nDeviceChannels[mode])
- stream_.doConvertBuffer[mode] = true;
- if (stream_.userFormat != stream_.deviceFormat[mode])
- stream_.doConvertBuffer[mode] = true;
- if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] &&
- stream_.nUserChannels[mode] > 1 )
- stream_.doConvertBuffer[mode] = true;
-
- // Allocate necessary internal buffers
- bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat );
- stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 );
- if ( stream_.userBuffer[mode] == NULL ) {
- errorText_ = "RtApiDs::probeDeviceOpen: error allocating user buffer memory.";
+ // Allocate our DsHandle structures for the stream.
+ if ( stream_.apiHandle == 0 ) {
+ try {
+ handle = new DsHandle;
+ }
+ catch ( std::bad_alloc& ) {
+ errorText_ = "RtApiDs::probeDeviceOpen: error allocating AsioHandle memory.";
goto error;
}
- if ( stream_.doConvertBuffer[mode] ) {
-
- bool makeBuffer = true;
- bufferBytes = stream_.nDeviceChannels[mode] * formatBytes( stream_.deviceFormat[mode] );
- if ( mode == INPUT ) {
- if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) {
- unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] );
- if ( bufferBytes <= (long) bytesOut ) makeBuffer = false;
- }
- }
-
- if ( makeBuffer ) {
- bufferBytes *= *bufferSize;
- if ( stream_.deviceBuffer ) free( stream_.deviceBuffer );
- stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 );
- if ( stream_.deviceBuffer == NULL ) {
- errorText_ = "RtApiDs::probeDeviceOpen: error allocating device buffer memory.";
- goto error;
- }
- }
- }
-
- // Allocate our DsHandle structures for the stream.
- if ( stream_.apiHandle == 0 ) {
- try {
- handle = new DsHandle;
- }
- catch ( std::bad_alloc& ) {
- errorText_ = "RtApiDs::probeDeviceOpen: error allocating AsioHandle memory.";
- goto error;
- }
-
- // Create a manual-reset event.
- handle->condition = CreateEvent( NULL, // no security
- TRUE, // manual-reset
- FALSE, // non-signaled initially
- NULL ); // unnamed
- stream_.apiHandle = (void *) handle;
- }
- else
- handle = (DsHandle *) stream_.apiHandle;
- handle->id[mode] = ohandle;
- handle->buffer[mode] = bhandle;
- handle->dsBufferSize[mode] = dsBufferSize;
- handle->dsPointerLeadTime[mode] = dsPointerLeadTime;
+ // Create a manual-reset event.
+ handle->condition = CreateEvent( NULL, // no security
+ TRUE, // manual-reset
+ FALSE, // non-signaled initially
+ NULL ); // unnamed
+ stream_.apiHandle = (void *) handle;
+ }
+ else
+ handle = (DsHandle *) stream_.apiHandle;
+ handle->id[mode] = ohandle;
+ handle->buffer[mode] = bhandle;
+ handle->dsBufferSize[mode] = dsBufferSize;
+ handle->dsPointerLeadTime[mode] = dsPointerLeadTime;
- stream_.device[mode] = device;
- stream_.state = STREAM_STOPPED;
- if ( stream_.mode == OUTPUT && mode == INPUT )
- // We had already set up an output stream.
- stream_.mode = DUPLEX;
- else
- stream_.mode = mode;
- stream_.nBuffers = nBuffers;
- stream_.sampleRate = sampleRate;
+ stream_.device[mode] = device;
+ stream_.state = STREAM_STOPPED;
+ if ( stream_.mode == OUTPUT && mode == INPUT )
+ // We had already set up an output stream.
+ stream_.mode = DUPLEX;
+ else
+ stream_.mode = mode;
+ stream_.nBuffers = nBuffers;
+ stream_.sampleRate = sampleRate;
- // Setup the buffer conversion information structure.
- if ( stream_.doConvertBuffer[mode] ) setConvertInfo( mode, firstChannel );
+ // Setup the buffer conversion information structure.
+ if ( stream_.doConvertBuffer[mode] ) setConvertInfo( mode, firstChannel );
- // Setup the callback thread.
- unsigned threadId;
- stream_.callbackInfo.object = (void *) this;
- stream_.callbackInfo.isRunning = true;
- stream_.callbackInfo.thread = _beginthreadex( NULL, 0, &callbackHandler,
- &stream_.callbackInfo, 0, &threadId );
- if ( stream_.callbackInfo.thread == 0 ) {
- errorText_ = "RtApiDs::probeDeviceOpen: error creating callback thread!";
- goto error;
- }
+ // Setup the callback thread.
+ unsigned threadId;
+ stream_.callbackInfo.object = (void *) this;
+ stream_.callbackInfo.isRunning = true;
+ stream_.callbackInfo.thread = _beginthreadex( NULL, 0, &callbackHandler,
+ &stream_.callbackInfo, 0, &threadId );
+ if ( stream_.callbackInfo.thread == 0 ) {
+ errorText_ = "RtApiDs::probeDeviceOpen: error creating callback thread!";
+ goto error;
+ }
- // Boost DS thread priority
- SetThreadPriority( (HANDLE) stream_.callbackInfo.thread, THREAD_PRIORITY_HIGHEST );
- return SUCCESS;
+ // Boost DS thread priority
+ SetThreadPriority( (HANDLE) stream_.callbackInfo.thread, THREAD_PRIORITY_HIGHEST );
+ return SUCCESS;
- error:
- if ( handle ) {
- if ( handle->buffer[0] ) { // the object pointer can be NULL and valid
- LPDIRECTSOUND object = (LPDIRECTSOUND) handle->id[0];
- LPDIRECTSOUNDBUFFER buffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0];
- if ( buffer ) buffer->Release();
- object->Release();
- }
- if ( handle->buffer[1] ) {
- LPDIRECTSOUNDCAPTURE object = (LPDIRECTSOUNDCAPTURE) handle->id[1];
- LPDIRECTSOUNDCAPTUREBUFFER buffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1];
- if ( buffer ) buffer->Release();
- object->Release();
- }
- CloseHandle( handle->condition );
- delete handle;
- stream_.apiHandle = 0;
+ error:
+ if ( handle ) {
+ if ( handle->buffer[0] ) { // the object pointer can be NULL and valid
+ LPDIRECTSOUND object = (LPDIRECTSOUND) handle->id[0];
+ LPDIRECTSOUNDBUFFER buffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0];
+ if ( buffer ) buffer->Release();
+ object->Release();
}
-
- for ( int i=0; i<2; i++ ) {
- if ( stream_.userBuffer[i] ) {
- free( stream_.userBuffer[i] );
- stream_.userBuffer[i] = 0;
- }
+ if ( handle->buffer[1] ) {
+ LPDIRECTSOUNDCAPTURE object = (LPDIRECTSOUNDCAPTURE) handle->id[1];
+ LPDIRECTSOUNDCAPTUREBUFFER buffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1];
+ if ( buffer ) buffer->Release();
+ object->Release();
}
+ CloseHandle( handle->condition );
+ delete handle;
+ stream_.apiHandle = 0;
+ }
- if ( stream_.deviceBuffer ) {
- free( stream_.deviceBuffer );
- stream_.deviceBuffer = 0;
+ for ( int i=0; i<2; i++ ) {
+ if ( stream_.userBuffer[i] ) {
+ free( stream_.userBuffer[i] );
+ stream_.userBuffer[i] = 0;
}
+ }
- return FAILURE;
+ if ( stream_.deviceBuffer ) {
+ free( stream_.deviceBuffer );
+ stream_.deviceBuffer = 0;
}
- void RtApiDs :: closeStream()
- {
- if ( stream_.state == STREAM_CLOSED ) {
- errorText_ = "RtApiDs::closeStream(): no open stream to close!";
- error( RtError::WARNING );
- return;
- }
+ return FAILURE;
+}
+
+void RtApiDs :: closeStream()
+{
+ if ( stream_.state == STREAM_CLOSED ) {
+ errorText_ = "RtApiDs::closeStream(): no open stream to close!";
+ error( RtError::WARNING );
+ return;
+ }
- // Stop the callback thread.
- stream_.callbackInfo.isRunning = false;
- WaitForSingleObject( (HANDLE) stream_.callbackInfo.thread, INFINITE );
- CloseHandle( (HANDLE) stream_.callbackInfo.thread );
+ // Stop the callback thread.
+ stream_.callbackInfo.isRunning = false;
+ WaitForSingleObject( (HANDLE) stream_.callbackInfo.thread, INFINITE );
+ CloseHandle( (HANDLE) stream_.callbackInfo.thread );
- DsHandle *handle = (DsHandle *) stream_.apiHandle;
- if ( handle ) {
- if ( handle->buffer[0] ) { // the object pointer can be NULL and valid
- LPDIRECTSOUND object = (LPDIRECTSOUND) handle->id[0];
- LPDIRECTSOUNDBUFFER buffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0];
- if ( buffer ) {
- buffer->Stop();
- buffer->Release();
- }
- object->Release();
- }
- if ( handle->buffer[1] ) {
- LPDIRECTSOUNDCAPTURE object = (LPDIRECTSOUNDCAPTURE) handle->id[1];
- LPDIRECTSOUNDCAPTUREBUFFER buffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1];
- if ( buffer ) {
- buffer->Stop();
- buffer->Release();
- }
- object->Release();
+ DsHandle *handle = (DsHandle *) stream_.apiHandle;
+ if ( handle ) {
+ if ( handle->buffer[0] ) { // the object pointer can be NULL and valid
+ LPDIRECTSOUND object = (LPDIRECTSOUND) handle->id[0];
+ LPDIRECTSOUNDBUFFER buffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0];
+ if ( buffer ) {
+ buffer->Stop();
+ buffer->Release();
}
- CloseHandle( handle->condition );
- delete handle;
- stream_.apiHandle = 0;
+ object->Release();
}
-
- for ( int i=0; i<2; i++ ) {
- if ( stream_.userBuffer[i] ) {
- free( stream_.userBuffer[i] );
- stream_.userBuffer[i] = 0;
+ if ( handle->buffer[1] ) {
+ LPDIRECTSOUNDCAPTURE object = (LPDIRECTSOUNDCAPTURE) handle->id[1];
+ LPDIRECTSOUNDCAPTUREBUFFER buffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1];
+ if ( buffer ) {
+ buffer->Stop();
+ buffer->Release();
}
+ object->Release();
}
+ CloseHandle( handle->condition );
+ delete handle;
+ stream_.apiHandle = 0;
+ }
- if ( stream_.deviceBuffer ) {
- free( stream_.deviceBuffer );
- stream_.deviceBuffer = 0;
+ for ( int i=0; i<2; i++ ) {
+ if ( stream_.userBuffer[i] ) {
+ free( stream_.userBuffer[i] );
+ stream_.userBuffer[i] = 0;
}
+ }
- stream_.mode = UNINITIALIZED;
- stream_.state = STREAM_CLOSED;
+ if ( stream_.deviceBuffer ) {
+ free( stream_.deviceBuffer );
+ stream_.deviceBuffer = 0;
}
- void RtApiDs :: startStream()
- {
- verifyStream();
- if ( stream_.state == STREAM_RUNNING ) {
- errorText_ = "RtApiDs::startStream(): the stream is already running!";
- error( RtError::WARNING );
- return;
- }
+ stream_.mode = UNINITIALIZED;
+ stream_.state = STREAM_CLOSED;
+}
- // Increase scheduler frequency on lesser windows (a side-effect of
- // increasing timer accuracy). On greater windows (Win2K or later),
- // this is already in effect.
+void RtApiDs :: startStream()
+{
+ verifyStream();
+ if ( stream_.state == STREAM_RUNNING ) {
+ errorText_ = "RtApiDs::startStream(): the stream is already running!";
+ error( RtError::WARNING );
+ return;
+ }
- MUTEX_LOCK( &stream_.mutex );
+ MUTEX_LOCK( &stream_.mutex );
- DsHandle *handle = (DsHandle *) stream_.apiHandle;
+ DsHandle *handle = (DsHandle *) stream_.apiHandle;
- timeBeginPeriod( 1 );
+ // Increase scheduler frequency on lesser windows (a side-effect of
+ // increasing timer accuracy). On greater windows (Win2K or later),
+ // this is already in effect.
+ timeBeginPeriod( 1 );
- /*
- memset( &statistics, 0, sizeof( statistics ) );
- statistics.sampleRate = stream_.sampleRate;
- statistics.writeDeviceBufferLeadBytes = handle->dsPointerLeadTime[0];
- */
+ buffersRolling = false;
+ duplexPrerollBytes = 0;
- buffersRolling = false;
- duplexPrerollBytes = 0;
-
- if ( stream_.mode == DUPLEX ) {
- // 0.5 seconds of silence in DUPLEX mode while the devices spin up and synchronize.
- duplexPrerollBytes = (int) ( 0.5 * stream_.sampleRate * formatBytes( stream_.deviceFormat[1] ) * stream_.nDeviceChannels[1] );
- }
+ if ( stream_.mode == DUPLEX ) {
+ // 0.5 seconds of silence in DUPLEX mode while the devices spin up and synchronize.
+ duplexPrerollBytes = (int) ( 0.5 * stream_.sampleRate * formatBytes( stream_.deviceFormat[1] ) * stream_.nDeviceChannels[1] );
+ }
- HRESULT result = 0;
- if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
- //statistics.outputFrameSize = formatBytes( stream_.deviceFormat[0] ) * stream_.nDeviceChannels[0];
+ HRESULT result = 0;
+ if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
- LPDIRECTSOUNDBUFFER buffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0];
- result = buffer->Play( 0, 0, DSBPLAY_LOOPING );
- if ( FAILED( result ) ) {
- errorStream_ << "RtApiDs::startStream: error (" << getErrorString( result ) << ") starting output buffer!";
- errorText_ = errorStream_.str();
- goto unlock;
- }
+ LPDIRECTSOUNDBUFFER buffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0];
+ result = buffer->Play( 0, 0, DSBPLAY_LOOPING );
+ if ( FAILED( result ) ) {
+ errorStream_ << "RtApiDs::startStream: error (" << getErrorString( result ) << ") starting output buffer!";
+ errorText_ = errorStream_.str();
+ goto unlock;
}
+ }
- if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {
- //statistics.inputFrameSize = formatBytes( stream_.deviceFormat[1]) * stream_.nDeviceChannels[1];
+ if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {
- LPDIRECTSOUNDCAPTUREBUFFER buffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1];
- result = buffer->Start( DSCBSTART_LOOPING );
- if ( FAILED( result ) ) {
- errorStream_ << "RtApiDs::startStream: error (" << getErrorString( result ) << ") starting input buffer!";
- errorText_ = errorStream_.str();
- goto unlock;
- }
+ LPDIRECTSOUNDCAPTUREBUFFER buffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1];
+ result = buffer->Start( DSCBSTART_LOOPING );
+ if ( FAILED( result ) ) {
+ errorStream_ << "RtApiDs::startStream: error (" << getErrorString( result ) << ") starting input buffer!";
+ errorText_ = errorStream_.str();
+ goto unlock;
}
-
- handle->drainCounter = 0;
- handle->internalDrain = false;
- stream_.state = STREAM_RUNNING;
-
- unlock:
- MUTEX_UNLOCK( &stream_.mutex );
-
- if ( FAILED( result ) ) error( RtError::SYSTEM_ERROR );
}
- void RtApiDs :: stopStream()
- {
- verifyStream();
- if ( stream_.state == STREAM_STOPPED ) {
- errorText_ = "RtApiDs::stopStream(): the stream is already stopped!";
- error( RtError::WARNING );
- return;
- }
-
- MUTEX_LOCK( &stream_.mutex );
+ handle->drainCounter = 0;
+ handle->internalDrain = false;
+ stream_.state = STREAM_RUNNING;
- HRESULT result = 0;
- LPVOID audioPtr;
- DWORD dataLen;
- DsHandle *handle = (DsHandle *) stream_.apiHandle;
- if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
- if ( handle->drainCounter == 0 ) {
- handle->drainCounter = 1;
- MUTEX_UNLOCK( &stream_.mutex );
- WaitForMultipleObjects( 1, &handle->condition, FALSE, INFINITE ); // block until signaled
- ResetEvent( handle->condition );
- MUTEX_LOCK( &stream_.mutex );
- }
+ unlock:
+ MUTEX_UNLOCK( &stream_.mutex );
- // Stop the buffer and clear memory
- LPDIRECTSOUNDBUFFER buffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0];
- result = buffer->Stop();
- if ( FAILED( result ) ) {
- errorStream_ << "RtApiDs::abortStream: error (" << getErrorString( result ) << ") stopping output buffer!";
- errorText_ = errorStream_.str();
- goto unlock;
- }
+ if ( FAILED( result ) ) error( RtError::SYSTEM_ERROR );
+}
- // Lock the buffer and clear it so that if we start to play again,
- // we won't have old data playing.
- result = buffer->Lock( 0, handle->dsBufferSize[0], &audioPtr, &dataLen, NULL, NULL, 0 );
- if ( FAILED( result ) ) {
- errorStream_ << "RtApiDs::abortStream: error (" << getErrorString( result ) << ") locking output buffer!";
- errorText_ = errorStream_.str();
- goto unlock;
- }
+void RtApiDs :: stopStream()
+{
+ verifyStream();
+ if ( stream_.state == STREAM_STOPPED ) {
+ errorText_ = "RtApiDs::stopStream(): the stream is already stopped!";
+ error( RtError::WARNING );
+ return;
+ }
- // Zero the DS buffer
- ZeroMemory( audioPtr, dataLen );
+ MUTEX_LOCK( &stream_.mutex );
- // Unlock the DS buffer
- result = buffer->Unlock( audioPtr, dataLen, NULL, 0 );
- if ( FAILED( result ) ) {
- errorStream_ << "RtApiDs::abortStream: error (" << getErrorString( result ) << ") unlocking output buffer!";
- errorText_ = errorStream_.str();
- goto unlock;
- }
+ if ( stream_.state == STREAM_STOPPED ) {
+ MUTEX_UNLOCK( &stream_.mutex );
+ return;
+ }
- // If we start playing again, we must begin at beginning of buffer.
- handle->bufferPointer[0] = 0;
+ HRESULT result = 0;
+ LPVOID audioPtr;
+ DWORD dataLen;
+ DsHandle *handle = (DsHandle *) stream_.apiHandle;
+ if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
+ if ( handle->drainCounter == 0 ) {
+ handle->drainCounter = 1;
+ MUTEX_UNLOCK( &stream_.mutex );
+ WaitForMultipleObjects( 1, &handle->condition, FALSE, INFINITE ); // block until signaled
+ ResetEvent( handle->condition );
+ MUTEX_LOCK( &stream_.mutex );
}
- if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {
- LPDIRECTSOUNDCAPTUREBUFFER buffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1];
- audioPtr = NULL;
- dataLen = 0;
-
- result = buffer->Stop();
- if ( FAILED( result ) ) {
- errorStream_ << "RtApiDs::abortStream: error (" << getErrorString( result ) << ") stopping input buffer!";
- errorText_ = errorStream_.str();
- goto unlock;
- }
-
- // Lock the buffer and clear it so that if we start to play again,
- // we won't have old data playing.
- result = buffer->Lock( 0, handle->dsBufferSize[1], &audioPtr, &dataLen, NULL, NULL, 0 );
- if ( FAILED( result ) ) {
- errorStream_ << "RtApiDs::abortStream: error (" << getErrorString( result ) << ") locking input buffer!";
- errorText_ = errorStream_.str();
- goto unlock;
- }
-
- // Zero the DS buffer
- ZeroMemory( audioPtr, dataLen );
-
- // Unlock the DS buffer
- result = buffer->Unlock( audioPtr, dataLen, NULL, 0 );
- if ( FAILED( result ) ) {
- errorStream_ << "RtApiDs::abortStream: error (" << getErrorString( result ) << ") unlocking input buffer!";
- errorText_ = errorStream_.str();
- goto unlock;
- }
+ // Stop the buffer and clear memory
+ LPDIRECTSOUNDBUFFER buffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0];
+ result = buffer->Stop();
+ if ( FAILED( result ) ) {
+ errorStream_ << "RtApiDs::stopStream: error (" << getErrorString( result ) << ") stopping output buffer!";
+ errorText_ = errorStream_.str();
+ goto unlock;
+ }
- // If we start recording again, we must begin at beginning of buffer.
- handle->bufferPointer[1] = 0;
+ // Lock the buffer and clear it so that if we start to play again,
+ // we won't have old data playing.
+ result = buffer->Lock( 0, handle->dsBufferSize[0], &audioPtr, &dataLen, NULL, NULL, 0 );
+ if ( FAILED( result ) ) {
+ errorStream_ << "RtApiDs::stopStream: error (" << getErrorString( result ) << ") locking output buffer!";
+ errorText_ = errorStream_.str();
+ goto unlock;
}
- unlock:
- timeEndPeriod( 1 ); // revert to normal scheduler frequency on lesser windows.
- stream_.state = STREAM_STOPPED;
- MUTEX_UNLOCK( &stream_.mutex );
- if ( FAILED( result ) ) error( RtError::SYSTEM_ERROR );
- }
+ // Zero the DS buffer
+ ZeroMemory( audioPtr, dataLen );
- void RtApiDs :: abortStream()
- {
- verifyStream();
- if ( stream_.state == STREAM_STOPPED ) {
- errorText_ = "RtApiDs::abortStream(): the stream is already stopped!";
- error( RtError::WARNING );
- return;
+ // Unlock the DS buffer
+ result = buffer->Unlock( audioPtr, dataLen, NULL, 0 );
+ if ( FAILED( result ) ) {
+ errorStream_ << "RtApiDs::stopStream: error (" << getErrorString( result ) << ") unlocking output buffer!";
+ errorText_ = errorStream_.str();
+ goto unlock;
}
- DsHandle *handle = (DsHandle *) stream_.apiHandle;
- handle->drainCounter = 1;
-
- stopStream();
+ // If we start playing again, we must begin at beginning of buffer.
+ handle->bufferPointer[0] = 0;
}
- void RtApiDs :: callbackEvent()
- {
- if ( stream_.state == STREAM_STOPPED ) {
- Sleep(50); // sleep 50 milliseconds
- return;
- }
+ if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {
+ LPDIRECTSOUNDCAPTUREBUFFER buffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1];
+ audioPtr = NULL;
+ dataLen = 0;
- if ( stream_.state == STREAM_CLOSED ) {
- errorText_ = "RtApiDs::callbackEvent(): the stream is closed ... this shouldn't happen!";
- error( RtError::WARNING );
- return;
+ result = buffer->Stop();
+ if ( FAILED( result ) ) {
+ errorStream_ << "RtApiDs::stopStream: error (" << getErrorString( result ) << ") stopping input buffer!";
+ errorText_ = errorStream_.str();
+ goto unlock;
}
- CallbackInfo *info = (CallbackInfo *) &stream_.callbackInfo;
- DsHandle *handle = (DsHandle *) stream_.apiHandle;
-
- // Check if we were draining the stream and signal is finished.
- if ( handle->drainCounter > stream_.nBuffers + 2 ) {
- if ( handle->internalDrain == false )
- SetEvent( handle->condition );
- else
- stopStream();
- return;
+ // Lock the buffer and clear it so that if we start to play again,
+ // we won't have old data playing.
+ result = buffer->Lock( 0, handle->dsBufferSize[1], &audioPtr, &dataLen, NULL, NULL, 0 );
+ if ( FAILED( result ) ) {
+ errorStream_ << "RtApiDs::stopStream: error (" << getErrorString( result ) << ") locking input buffer!";
+ errorText_ = errorStream_.str();
+ goto unlock;
}
- MUTEX_LOCK( &stream_.mutex );
+ // Zero the DS buffer
+ ZeroMemory( audioPtr, dataLen );
- // Invoke user callback to get fresh output data UNLESS we are
- // draining stream.
- if ( handle->drainCounter == 0 ) {
- RtAudioCallback callback = (RtAudioCallback) info->callback;
- double streamTime = getStreamTime();
- RtAudioStreamStatus status = 0;
- if ( stream_.mode != INPUT && handle->xrun[0] == true ) {
- status |= RTAUDIO_OUTPUT_UNDERFLOW;
- handle->xrun[0] = false;
- }
- if ( stream_.mode != OUTPUT && handle->xrun[1] == true ) {
- status |= RTAUDIO_INPUT_OVERFLOW;
- handle->xrun[1] = false;
- }
- handle->drainCounter = callback( stream_.userBuffer[0], stream_.userBuffer[1],
- stream_.bufferSize, streamTime, status, info->userData );
- if ( handle->drainCounter == 2 ) {
- MUTEX_UNLOCK( &stream_.mutex );
- abortStream();
- return;
- }
- else if ( handle->drainCounter == 1 )
- handle->internalDrain = true;
+ // Unlock the DS buffer
+ result = buffer->Unlock( audioPtr, dataLen, NULL, 0 );
+ if ( FAILED( result ) ) {
+ errorStream_ << "RtApiDs::stopStream: error (" << getErrorString( result ) << ") unlocking input buffer!";
+ errorText_ = errorStream_.str();
+ goto unlock;
}
- HRESULT result;
- DWORD currentWritePos, safeWritePos;
- DWORD currentReadPos, safeReadPos;
- DWORD leadPos;
- UINT nextWritePos;
+ // If we start recording again, we must begin at beginning of buffer.
+ handle->bufferPointer[1] = 0;
+ }
-#ifdef GENERATE_DEBUG_LOG
- DWORD writeTime, readTime;
-#endif
+ unlock:
+ timeEndPeriod( 1 ); // revert to normal scheduler frequency on lesser windows.
+ stream_.state = STREAM_STOPPED;
+ MUTEX_UNLOCK( &stream_.mutex );
- LPVOID buffer1 = NULL;
- LPVOID buffer2 = NULL;
- DWORD bufferSize1 = 0;
- DWORD bufferSize2 = 0;
+ if ( FAILED( result ) ) error( RtError::SYSTEM_ERROR );
+}
- char *buffer;
- long bufferBytes;
+void RtApiDs :: abortStream()
+{
+ verifyStream();
+ if ( stream_.state == STREAM_STOPPED ) {
+ errorText_ = "RtApiDs::abortStream(): the stream is already stopped!";
+ error( RtError::WARNING );
+ return;
+ }
- if ( stream_.mode == DUPLEX && !buffersRolling ) {
- assert( handle->dsBufferSize[0] == handle->dsBufferSize[1] );
+ DsHandle *handle = (DsHandle *) stream_.apiHandle;
+ handle->drainCounter = 1;
- // It takes a while for the devices to get rolling. As a result,
- // there's no guarantee that the capture and write device pointers
- // will move in lockstep. Wait here for both devices to start
- // rolling, and then set our buffer pointers accordingly.
- // e.g. Crystal Drivers: the capture buffer starts up 5700 to 9600
- // bytes later than the write buffer.
+ stopStream();
+}
- // Stub: a serious risk of having a pre-emptive scheduling round
- // take place between the two GetCurrentPosition calls... but I'm
+void RtApiDs :: callbackEvent()
+{
+ if ( stream_.state == STREAM_STOPPED ) {
+ Sleep( 50 ); // sleep 50 milliseconds
+ return;
+ }
+
+ if ( stream_.state == STREAM_CLOSED ) {
+ errorText_ = "RtApiDs::callbackEvent(): the stream is closed ... this shouldn't happen!";
+ error( RtError::WARNING );
+ return;
+ }
+
+ CallbackInfo *info = (CallbackInfo *) &stream_.callbackInfo;
+ DsHandle *handle = (DsHandle *) stream_.apiHandle;
+
+ // Check if we were draining the stream and signal is finished.
+ if ( handle->drainCounter > stream_.nBuffers + 2 ) {
+ if ( handle->internalDrain == false )
+ SetEvent( handle->condition );
+ else
+ stopStream();
+ return;
+ }
+
+ MUTEX_LOCK( &stream_.mutex );
+
+ // The state might change while waiting on a mutex.
+ if ( stream_.state == STREAM_STOPPED ) {
+ MUTEX_UNLOCK( &stream_.mutex );
+ return;
+ }
+
+ // Invoke user callback to get fresh output data UNLESS we are
+ // draining stream.
+ if ( handle->drainCounter == 0 ) {
+ RtAudioCallback callback = (RtAudioCallback) info->callback;
+ double streamTime = getStreamTime();
+ RtAudioStreamStatus status = 0;
+ if ( stream_.mode != INPUT && handle->xrun[0] == true ) {
+ status |= RTAUDIO_OUTPUT_UNDERFLOW;
+ handle->xrun[0] = false;
+ }
+ if ( stream_.mode != OUTPUT && handle->xrun[1] == true ) {
+ status |= RTAUDIO_INPUT_OVERFLOW;
+ handle->xrun[1] = false;
+ }
+ handle->drainCounter = callback( stream_.userBuffer[0], stream_.userBuffer[1],
+ stream_.bufferSize, streamTime, status, info->userData );
+ if ( handle->drainCounter == 2 ) {
+ MUTEX_UNLOCK( &stream_.mutex );
+ abortStream();
+ return;
+ }
+ else if ( handle->drainCounter == 1 )
+ handle->internalDrain = true;
+ }
+
+ HRESULT result;
+ DWORD currentWritePointer, safeWritePointer;
+ DWORD currentReadPointer, safeReadPointer;
+ UINT nextWritePointer;
+
+ LPVOID buffer1 = NULL;
+ LPVOID buffer2 = NULL;
+ DWORD bufferSize1 = 0;
+ DWORD bufferSize2 = 0;
+
+ char *buffer;
+ long bufferBytes;
+
+ if ( buffersRolling == false ) {
+ if ( stream_.mode == DUPLEX ) {
+ //assert( handle->dsBufferSize[0] == handle->dsBufferSize[1] );
+
+ // It takes a while for the devices to get rolling. As a result,
+ // there's no guarantee that the capture and write device pointers
+ // will move in lockstep. Wait here for both devices to start
+ // rolling, and then set our buffer pointers accordingly.
+ // e.g. Crystal Drivers: the capture buffer starts up 5700 to 9600
+ // bytes later than the write buffer.
+
+ // Stub: a serious risk of having a pre-emptive scheduling round
+ // take place between the two GetCurrentPosition calls... but I'm
// really not sure how to solve the problem. Temporarily boost to
// Realtime priority, maybe; but I'm not sure what priority the
// DirectSound service threads run at. We *should* be roughly
LPDIRECTSOUNDBUFFER dsWriteBuffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0];
LPDIRECTSOUNDCAPTUREBUFFER dsCaptureBuffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1];
- DWORD initialWritePos, initialSafeWritePos;
- DWORD initialReadPos, initialSafeReadPos;
+ DWORD startSafeWritePointer, startSafeReadPointer;
- result = dsWriteBuffer->GetCurrentPosition( &initialWritePos, &initialSafeWritePos );
+ result = dsWriteBuffer->GetCurrentPosition( NULL, &startSafeWritePointer );
if ( FAILED( result ) ) {
errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current write position!";
errorText_ = errorStream_.str();
error( RtError::SYSTEM_ERROR );
}
- result = dsCaptureBuffer->GetCurrentPosition( &initialReadPos, &initialSafeReadPos );
+ result = dsCaptureBuffer->GetCurrentPosition( NULL, &startSafeReadPointer );
if ( FAILED( result ) ) {
errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current read position!";
errorText_ = errorStream_.str();
error( RtError::SYSTEM_ERROR );
}
while ( true ) {
- result = dsWriteBuffer->GetCurrentPosition( ¤tWritePos, &safeWritePos );
+ result = dsWriteBuffer->GetCurrentPosition( NULL, &safeWritePointer );
if ( FAILED( result ) ) {
errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current write position!";
errorText_ = errorStream_.str();
error( RtError::SYSTEM_ERROR );
}
- result = dsCaptureBuffer->GetCurrentPosition( ¤tReadPos, &safeReadPos );
+ result = dsCaptureBuffer->GetCurrentPosition( NULL, &safeReadPointer );
if ( FAILED( result ) ) {
errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current read position!";
errorText_ = errorStream_.str();
error( RtError::SYSTEM_ERROR );
}
- if ( safeWritePos != initialSafeWritePos && safeReadPos != initialSafeReadPos ) break;
+ if ( safeWritePointer != startSafeWritePointer && safeReadPointer != startSafeReadPointer ) break;
Sleep( 1 );
}
- assert( handle->dsBufferSize[0] == handle->dsBufferSize[1] );
+ //assert( handle->dsBufferSize[0] == handle->dsBufferSize[1] );
- buffersRolling = true;
- handle->bufferPointer[0] = ( safeWritePos + handle->dsPointerLeadTime[0] );
- handle->bufferPointer[1] = safeReadPos;
+ handle->bufferPointer[0] = safeWritePointer + handle->dsPointerLeadTime[0];
+ if ( handle->bufferPointer[0] >= handle->dsBufferSize[0] ) handle->bufferPointer[0] -= handle->dsBufferSize[0];
+ handle->bufferPointer[1] = safeReadPointer;
}
+ else if ( stream_.mode == OUTPUT ) {
- if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
-
- LPDIRECTSOUNDBUFFER dsBuffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0];
-
- if ( handle->drainCounter > 1 ) { // write zeros to the output stream
- bufferBytes = stream_.bufferSize * stream_.nUserChannels[0];
- bufferBytes *= formatBytes( stream_.userFormat );
- memset( stream_.userBuffer[0], 0, bufferBytes );
- }
-
- // Setup parameters and do buffer conversion if necessary.
- if ( stream_.doConvertBuffer[0] ) {
- buffer = stream_.deviceBuffer;
- convertBuffer( buffer, stream_.userBuffer[0], stream_.convertInfo[0] );
- bufferBytes = stream_.bufferSize * stream_.nDeviceChannels[0];
- bufferBytes *= formatBytes( stream_.deviceFormat[0] );
- }
- else {
- buffer = stream_.userBuffer[0];
- bufferBytes = stream_.bufferSize * stream_.nUserChannels[0];
- bufferBytes *= formatBytes( stream_.userFormat );
+ // Set the proper nextWritePosition after initial startup.
+ LPDIRECTSOUNDBUFFER dsWriteBuffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0];
+ result = dsWriteBuffer->GetCurrentPosition( ¤tWritePointer, &safeWritePointer );
+ if ( FAILED( result ) ) {
+ errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current write position!";
+ errorText_ = errorStream_.str();
+ error( RtError::SYSTEM_ERROR );
}
+ handle->bufferPointer[0] = safeWritePointer + handle->dsPointerLeadTime[0];
+ if ( handle->bufferPointer[0] >= handle->dsBufferSize[0] ) handle->bufferPointer[0] -= handle->dsBufferSize[0];
+ }
- // No byte swapping necessary in DirectSound implementation.
+ buffersRolling = true;
+ }
- // Ahhh ... windoze. 16-bit data is signed but 8-bit data is
- // unsigned. So, we need to convert our signed 8-bit data here to
- // unsigned.
- if ( stream_.deviceFormat[0] == RTAUDIO_SINT8 )
- for ( int i=0; i<bufferBytes; i++ ) buffer[i] = (unsigned char) ( buffer[i] + 128 );
+ if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
+
+ LPDIRECTSOUNDBUFFER dsBuffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0];
- DWORD dsBufferSize = handle->dsBufferSize[0];
- nextWritePos = handle->bufferPointer[0];
+ if ( handle->drainCounter > 1 ) { // write zeros to the output stream
+ bufferBytes = stream_.bufferSize * stream_.nUserChannels[0];
+ bufferBytes *= formatBytes( stream_.userFormat );
+ memset( stream_.userBuffer[0], 0, bufferBytes );
+ }
- DWORD endWrite;
- while ( true ) {
- // Find out where the read and "safe write" pointers are.
- result = dsBuffer->GetCurrentPosition( ¤tWritePos, &safeWritePos );
- if ( FAILED( result ) ) {
- errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current write position!";
- errorText_ = errorStream_.str();
- error( RtError::SYSTEM_ERROR );
- }
+ // Setup parameters and do buffer conversion if necessary.
+ if ( stream_.doConvertBuffer[0] ) {
+ buffer = stream_.deviceBuffer;
+ convertBuffer( buffer, stream_.userBuffer[0], stream_.convertInfo[0] );
+ bufferBytes = stream_.bufferSize * stream_.nDeviceChannels[0];
+ bufferBytes *= formatBytes( stream_.deviceFormat[0] );
+ }
+ else {
+ buffer = stream_.userBuffer[0];
+ bufferBytes = stream_.bufferSize * stream_.nUserChannels[0];
+ bufferBytes *= formatBytes( stream_.userFormat );
+ }
- leadPos = safeWritePos + handle->dsPointerLeadTime[0];
- if ( leadPos > dsBufferSize ) leadPos -= dsBufferSize;
- if ( leadPos < nextWritePos ) leadPos += dsBufferSize; // unwrap offset
- endWrite = nextWritePos + bufferBytes;
-
- // Check whether the entire write region is behind the play pointer.
- if ( leadPos >= endWrite ) break;
-
- // If we are here, then we must wait until the play pointer gets
- // beyond the write region. The approach here is to use the
- // Sleep() function to suspend operation until safePos catches
- // up. Calculate number of milliseconds to wait as:
- // time = distance * (milliseconds/second) * fudgefactor /
- // ((bytes/sample) * (samples/second))
- // A "fudgefactor" less than 1 is used because it was found
- // that sleeping too long was MUCH worse than sleeping for
- // several shorter periods.
- double millis = ( endWrite - leadPos ) * 900.0;
- millis /= ( formatBytes( stream_.deviceFormat[0]) * stream_.nDeviceChannels[0] * stream_.sampleRate);
- if ( millis < 1.0 ) millis = 1.0;
- if ( millis > 50.0 ) {
- static int nOverruns = 0;
- ++nOverruns;
- }
- Sleep( (DWORD) millis );
- }
+ // No byte swapping necessary in DirectSound implementation.
- //if ( statistics.writeDeviceSafeLeadBytes < dsPointerDifference( safeWritePos, currentWritePos, handle->dsBufferSize[0] ) ) {
- // statistics.writeDeviceSafeLeadBytes = dsPointerDifference( safeWritePos, currentWritePos, handle->dsBufferSize[0] );
- //}
+ // Ahhh ... windoze. 16-bit data is signed but 8-bit data is
+ // unsigned. So, we need to convert our signed 8-bit data here to
+ // unsigned.
+ if ( stream_.deviceFormat[0] == RTAUDIO_SINT8 )
+ for ( int i=0; i<bufferBytes; i++ ) buffer[i] = (unsigned char) ( buffer[i] + 128 );
- if ( dsPointerBetween( nextWritePos, safeWritePos, currentWritePos, dsBufferSize )
- || dsPointerBetween( endWrite, safeWritePos, currentWritePos, dsBufferSize ) ) {
- // We've strayed into the forbidden zone ... resync the read pointer.
- //++statistics.numberOfWriteUnderruns;
- handle->xrun[0] = true;
- nextWritePos = safeWritePos + handle->dsPointerLeadTime[0] - bufferBytes + dsBufferSize;
- while ( nextWritePos >= dsBufferSize ) nextWritePos -= dsBufferSize;
- handle->bufferPointer[0] = nextWritePos;
- endWrite = nextWritePos + bufferBytes;
- }
+ DWORD dsBufferSize = handle->dsBufferSize[0];
+ nextWritePointer = handle->bufferPointer[0];
- // Lock free space in the buffer
- result = dsBuffer->Lock( nextWritePos, bufferBytes, &buffer1,
- &bufferSize1, &buffer2, &bufferSize2, 0 );
+ DWORD endWrite, leadPointer;
+ while ( true ) {
+ // Find out where the read and "safe write" pointers are.
+ result = dsBuffer->GetCurrentPosition( ¤tWritePointer, &safeWritePointer );
if ( FAILED( result ) ) {
- errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") locking buffer during playback!";
+ errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current write position!";
errorText_ = errorStream_.str();
error( RtError::SYSTEM_ERROR );
}
- // Copy our buffer into the DS buffer
- CopyMemory( buffer1, buffer, bufferSize1 );
- if ( buffer2 != NULL ) CopyMemory( buffer2, buffer+bufferSize1, bufferSize2 );
+ // We will copy our output buffer into the region between
+ // safeWritePointer and leadPointer. If leadPointer is not
+ // beyond the next endWrite position, wait until it is.
+ leadPointer = safeWritePointer + handle->dsPointerLeadTime[0];
+ //std::cout << "safeWritePointer = " << safeWritePointer << ", leadPointer = " << leadPointer << ", nextWritePointer = " << nextWritePointer << std::endl;
+ if ( leadPointer > dsBufferSize ) leadPointer -= dsBufferSize;
+ if ( leadPointer < nextWritePointer ) leadPointer += dsBufferSize; // unwrap offset
+ endWrite = nextWritePointer + bufferBytes;
- // Update our buffer offset and unlock sound buffer
- dsBuffer->Unlock( buffer1, bufferSize1, buffer2, bufferSize2 );
- if ( FAILED( result ) ) {
- errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") unlocking buffer during playback!";
- errorText_ = errorStream_.str();
- error( RtError::SYSTEM_ERROR );
- }
- nextWritePos = ( nextWritePos + bufferSize1 + bufferSize2 ) % dsBufferSize;
- handle->bufferPointer[0] = nextWritePos;
+ // Check whether the entire write region is behind the play pointer.
+ if ( leadPointer >= endWrite ) break;
- if ( handle->drainCounter ) {
- handle->drainCounter++;
- goto unlock;
- }
+ // If we are here, then we must wait until the leadPointer advances
+ // beyond the end of our next write region. We use the
+ // Sleep() function to suspend operation until that happens.
+ double millis = ( endWrite - leadPointer ) * 1000.0;
+ millis /= ( formatBytes( stream_.deviceFormat[0]) * stream_.nDeviceChannels[0] * stream_.sampleRate);
+ if ( millis < 1.0 ) millis = 1.0;
+ Sleep( (DWORD) millis );
}
- if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {
+ if ( dsPointerBetween( nextWritePointer, safeWritePointer, currentWritePointer, dsBufferSize )
+ || dsPointerBetween( endWrite, safeWritePointer, currentWritePointer, dsBufferSize ) ) {
+ // We've strayed into the forbidden zone ... resync the read pointer.
+ handle->xrun[0] = true;
+ nextWritePointer = safeWritePointer + handle->dsPointerLeadTime[0] - bufferBytes;
+ if ( nextWritePointer >= dsBufferSize ) nextWritePointer -= dsBufferSize;
+ handle->bufferPointer[0] = nextWritePointer;
+ endWrite = nextWritePointer + bufferBytes;
+ }
- // Setup parameters.
- if ( stream_.doConvertBuffer[1] ) {
- buffer = stream_.deviceBuffer;
- bufferBytes = stream_.bufferSize * stream_.nDeviceChannels[1];
- bufferBytes *= formatBytes( stream_.deviceFormat[1] );
- }
- else {
- buffer = stream_.userBuffer[1];
- bufferBytes = stream_.bufferSize * stream_.nUserChannels[1];
- bufferBytes *= formatBytes( stream_.userFormat );
- }
+ // Lock free space in the buffer
+ result = dsBuffer->Lock( nextWritePointer, bufferBytes, &buffer1,
+ &bufferSize1, &buffer2, &bufferSize2, 0 );
+ if ( FAILED( result ) ) {
+ errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") locking buffer during playback!";
+ errorText_ = errorStream_.str();
+ error( RtError::SYSTEM_ERROR );
+ }
- LPDIRECTSOUNDCAPTUREBUFFER dsBuffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1];
- long nextReadPos = handle->bufferPointer[1];
- DWORD dsBufferSize = handle->dsBufferSize[1];
+ // Copy our buffer into the DS buffer
+ CopyMemory( buffer1, buffer, bufferSize1 );
+ if ( buffer2 != NULL ) CopyMemory( buffer2, buffer+bufferSize1, bufferSize2 );
- // Find out where the write and "safe read" pointers are.
- result = dsBuffer->GetCurrentPosition( ¤tReadPos, &safeReadPos );
- if ( FAILED( result ) ) {
- errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current read position!";
- errorText_ = errorStream_.str();
- error( RtError::SYSTEM_ERROR );
- }
+ // Update our buffer offset and unlock sound buffer
+ dsBuffer->Unlock( buffer1, bufferSize1, buffer2, bufferSize2 );
+ if ( FAILED( result ) ) {
+ errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") unlocking buffer during playback!";
+ errorText_ = errorStream_.str();
+ error( RtError::SYSTEM_ERROR );
+ }
+ nextWritePointer = ( nextWritePointer + bufferSize1 + bufferSize2 ) % dsBufferSize;
+ handle->bufferPointer[0] = nextWritePointer;
- if ( safeReadPos < (DWORD)nextReadPos ) safeReadPos += dsBufferSize; // unwrap offset
- DWORD endRead = nextReadPos + bufferBytes;
-
- // Handling depends on whether we are INPUT or DUPLEX.
- // If we're in INPUT mode then waiting is a good thing. If we're in DUPLEX mode,
- // then a wait here will drag the write pointers into the forbidden zone.
- //
- // In DUPLEX mode, rather than wait, we will back off the read pointer until
- // it's in a safe position. This causes dropouts, but it seems to be the only
- // practical way to sync up the read and write pointers reliably, given the
- // the very complex relationship between phase and increment of the read and write
- // pointers.
- //
- // In order to minimize audible dropouts in DUPLEX mode, we will
- // provide a pre-roll period of 0.5 seconds in which we return
- // zeros from the read buffer while the pointers sync up.
-
- if ( stream_.mode == DUPLEX ) {
- if ( safeReadPos < endRead ) {
- if ( duplexPrerollBytes <= 0 ) {
- // Pre-roll time over. Be more agressive.
- int adjustment = endRead-safeReadPos;
-
- handle->xrun[1] = true;
- //++statistics.numberOfReadOverruns;
- // Two cases:
- // - large adjustments: we've probably run out of CPU cycles, so just resync exactly,
- // and perform fine adjustments later.
- // - small adjustments: back off by twice as much.
- if ( adjustment >= 2*bufferBytes )
- nextReadPos = safeReadPos-2*bufferBytes;
- else
- nextReadPos = safeReadPos-bufferBytes-adjustment;
-
- //statistics.readDeviceSafeLeadBytes = currentReadPos-nextReadPos;
- //if ( statistics.readDeviceSafeLeadBytes < 0) statistics.readDeviceSafeLeadBytes += dsBufferSize;
- if ( nextReadPos < 0 ) nextReadPos += dsBufferSize;
+ if ( handle->drainCounter ) {
+ handle->drainCounter++;
+ goto unlock;
+ }
+ }
- }
- else {
- // In pre=roll time. Just do it.
- nextReadPos = safeReadPos-bufferBytes;
- while ( nextReadPos < 0 ) nextReadPos += dsBufferSize;
- }
- endRead = nextReadPos + bufferBytes;
+ if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {
+
+ // Setup parameters.
+ if ( stream_.doConvertBuffer[1] ) {
+ buffer = stream_.deviceBuffer;
+ bufferBytes = stream_.bufferSize * stream_.nDeviceChannels[1];
+ bufferBytes *= formatBytes( stream_.deviceFormat[1] );
+ }
+ else {
+ buffer = stream_.userBuffer[1];
+ bufferBytes = stream_.bufferSize * stream_.nUserChannels[1];
+ bufferBytes *= formatBytes( stream_.userFormat );
+ }
+
+ LPDIRECTSOUNDCAPTUREBUFFER dsBuffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1];
+ long nextReadPointer = handle->bufferPointer[1];
+ DWORD dsBufferSize = handle->dsBufferSize[1];
+
+ // Find out where the write and "safe read" pointers are.
+ result = dsBuffer->GetCurrentPosition( ¤tReadPointer, &safeReadPointer );
+ if ( FAILED( result ) ) {
+ errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current read position!";
+ errorText_ = errorStream_.str();
+ error( RtError::SYSTEM_ERROR );
+ }
+
+ if ( safeReadPointer < (DWORD)nextReadPointer ) safeReadPointer += dsBufferSize; // unwrap offset
+ DWORD endRead = nextReadPointer + bufferBytes;
+
+ // Handling depends on whether we are INPUT or DUPLEX.
+ // If we're in INPUT mode then waiting is a good thing. If we're in DUPLEX mode,
+ // then a wait here will drag the write pointers into the forbidden zone.
+ //
+ // In DUPLEX mode, rather than wait, we will back off the read pointer until
+ // it's in a safe position. This causes dropouts, but it seems to be the only
+ // practical way to sync up the read and write pointers reliably, given the
+ // the very complex relationship between phase and increment of the read and write
+ // pointers.
+ //
+ // In order to minimize audible dropouts in DUPLEX mode, we will
+ // provide a pre-roll period of 0.5 seconds in which we return
+ // zeros from the read buffer while the pointers sync up.
+
+ if ( stream_.mode == DUPLEX ) {
+ if ( safeReadPointer < endRead ) {
+ if ( duplexPrerollBytes <= 0 ) {
+ // Pre-roll time over. Be more agressive.
+ int adjustment = endRead-safeReadPointer;
+
+ handle->xrun[1] = true;
+ // Two cases:
+ // - large adjustments: we've probably run out of CPU cycles, so just resync exactly,
+ // and perform fine adjustments later.
+ // - small adjustments: back off by twice as much.
+ if ( adjustment >= 2*bufferBytes )
+ nextReadPointer = safeReadPointer-2*bufferBytes;
+ else
+ nextReadPointer = safeReadPointer-bufferBytes-adjustment;
+
+ if ( nextReadPointer < 0 ) nextReadPointer += dsBufferSize;
+
+ }
+ else {
+ // In pre=roll time. Just do it.
+ nextReadPointer = safeReadPointer - bufferBytes;
+ while ( nextReadPointer < 0 ) nextReadPointer += dsBufferSize;
}
+ endRead = nextReadPointer + bufferBytes;
}
- else { // mode == INPUT
- while ( safeReadPos < endRead ) {
- // See comments for playback.
- double millis = (endRead - safeReadPos) * 900.0;
- millis /= ( formatBytes(stream_.deviceFormat[1]) * stream_.nDeviceChannels[1] * stream_.sampleRate);
- if ( millis < 1.0 ) millis = 1.0;
- Sleep( (DWORD) millis );
-
- // Wake up, find out where we are now
- result = dsBuffer->GetCurrentPosition( ¤tReadPos, &safeReadPos );
- if ( FAILED( result ) ) {
- errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current read position!";
- errorText_ = errorStream_.str();
- error( RtError::SYSTEM_ERROR );
- }
-
- if ( safeReadPos < (DWORD)nextReadPos ) safeReadPos += dsBufferSize; // unwrap offset
+ }
+ else { // mode == INPUT
+ while ( safeReadPointer < endRead ) {
+ // See comments for playback.
+ double millis = (endRead - safeReadPointer) * 1000.0;
+ millis /= ( formatBytes(stream_.deviceFormat[1]) * stream_.nDeviceChannels[1] * stream_.sampleRate);
+ if ( millis < 1.0 ) millis = 1.0;
+ Sleep( (DWORD) millis );
+
+ // Wake up and find out where we are now.
+ result = dsBuffer->GetCurrentPosition( ¤tReadPointer, &safeReadPointer );
+ if ( FAILED( result ) ) {
+ errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current read position!";
+ errorText_ = errorStream_.str();
+ error( RtError::SYSTEM_ERROR );
}
+
+ if ( safeReadPointer < (DWORD)nextReadPointer ) safeReadPointer += dsBufferSize; // unwrap offset
}
+ }
- //if (statistics.readDeviceSafeLeadBytes < dsPointerDifference( currentReadPos, nextReadPos, dsBufferSize ) )
- // statistics.readDeviceSafeLeadBytes = dsPointerDifference( currentReadPos, nextReadPos, dsBufferSize );
+ // Lock free space in the buffer
+ result = dsBuffer->Lock( nextReadPointer, bufferBytes, &buffer1,
+ &bufferSize1, &buffer2, &bufferSize2, 0 );
+ if ( FAILED( result ) ) {
+ errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") locking capture buffer!";
+ errorText_ = errorStream_.str();
+ error( RtError::SYSTEM_ERROR );
+ }
- // Lock free space in the buffer
- result = dsBuffer->Lock( nextReadPos, bufferBytes, &buffer1,
- &bufferSize1, &buffer2, &bufferSize2, 0 );
- if ( FAILED( result ) ) {
- errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") locking capture buffer!";
- errorText_ = errorStream_.str();
- error( RtError::SYSTEM_ERROR );
- }
+ if ( duplexPrerollBytes <= 0 ) {
+ // Copy our buffer into the DS buffer
+ CopyMemory( buffer, buffer1, bufferSize1 );
+ if ( buffer2 != NULL ) CopyMemory( buffer+bufferSize1, buffer2, bufferSize2 );
+ }
+ else {
+ memset( buffer, 0, bufferSize1 );
+ if ( buffer2 != NULL ) memset( buffer + bufferSize1, 0, bufferSize2 );
+ duplexPrerollBytes -= bufferSize1 + bufferSize2;
+ }
- if ( duplexPrerollBytes <= 0 ) {
- // Copy our buffer into the DS buffer
- CopyMemory( buffer, buffer1, bufferSize1 );
- if ( buffer2 != NULL ) CopyMemory( buffer+bufferSize1, buffer2, bufferSize2 );
- }
- else {
- memset( buffer, 0, bufferSize1 );
- if ( buffer2 != NULL ) memset( buffer + bufferSize1, 0, bufferSize2 );
- duplexPrerollBytes -= bufferSize1 + bufferSize2;
- }
+ // Update our buffer offset and unlock sound buffer
+ nextReadPointer = ( nextReadPointer + bufferSize1 + bufferSize2 ) % dsBufferSize;
+ dsBuffer->Unlock( buffer1, bufferSize1, buffer2, bufferSize2 );
+ if ( FAILED( result ) ) {
+ errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") unlocking capture buffer!";
+ errorText_ = errorStream_.str();
+ error( RtError::SYSTEM_ERROR );
+ }
+ handle->bufferPointer[1] = nextReadPointer;
- // Update our buffer offset and unlock sound buffer
- nextReadPos = ( nextReadPos + bufferSize1 + bufferSize2 ) % dsBufferSize;
- dsBuffer->Unlock( buffer1, bufferSize1, buffer2, bufferSize2 );
- if ( FAILED( result ) ) {
- errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") unlocking capture buffer!";
- errorText_ = errorStream_.str();
- error( RtError::SYSTEM_ERROR );
- }
- handle->bufferPointer[1] = nextReadPos;
-
- // No byte swapping necessary in DirectSound implementation.
-
- // If necessary, convert 8-bit data from unsigned to signed.
- if ( stream_.deviceFormat[1] == RTAUDIO_SINT8 )
- for ( int j=0; j<bufferBytes; j++ ) buffer[j] = (signed char) ( buffer[j] - 128 );
-
- // Do buffer conversion if necessary.
- if ( stream_.doConvertBuffer[1] )
- convertBuffer( stream_.userBuffer[1], stream_.deviceBuffer, stream_.convertInfo[1] );
- }
-#ifdef GENERATE_DEBUG_LOG
- if ( currentDebugLogEntry < debugLog.size() )
- {
- TTickRecord &r = debugLog[currentDebugLogEntry++];
- r.currentReadPointer = currentReadPos;
- r.safeReadPointer = safeReadPos;
- r.currentWritePointer = currentWritePos;
- r.safeWritePointer = safeWritePos;
- r.readTime = readTime;
- r.writeTime = writeTime;
- r.nextReadPointer = handles[1].bufferPointer;
- r.nextWritePointer = handles[0].bufferPointer;
- }
-#endif
+ // No byte swapping necessary in DirectSound implementation.
- unlock:
- MUTEX_UNLOCK( &stream_.mutex );
+ // If necessary, convert 8-bit data from unsigned to signed.
+ if ( stream_.deviceFormat[1] == RTAUDIO_SINT8 )
+ for ( int j=0; j<bufferBytes; j++ ) buffer[j] = (signed char) ( buffer[j] - 128 );
- RtApi::tickStreamTime();
+ // Do buffer conversion if necessary.
+ if ( stream_.doConvertBuffer[1] )
+ convertBuffer( stream_.userBuffer[1], stream_.deviceBuffer, stream_.convertInfo[1] );
}
- // Definitions for utility functions and callbacks
- // specific to the DirectSound implementation.
+ unlock:
+ MUTEX_UNLOCK( &stream_.mutex );
- extern "C" unsigned __stdcall callbackHandler( void *ptr )
- {
- CallbackInfo *info = (CallbackInfo *) ptr;
- RtApiDs *object = (RtApiDs *) info->object;
- bool* isRunning = &info->isRunning;
+ RtApi::tickStreamTime();
+}
- while ( *isRunning == true ) {
- object->callbackEvent();
- }
+// Definitions for utility functions and callbacks
+// specific to the DirectSound implementation.
- _endthreadex( 0 );
- return 0;
+extern "C" unsigned __stdcall callbackHandler( void *ptr )
+{
+ CallbackInfo *info = (CallbackInfo *) ptr;
+ RtApiDs *object = (RtApiDs *) info->object;
+ bool* isRunning = &info->isRunning;
+
+ while ( *isRunning == true ) {
+ object->callbackEvent();
}
+ _endthreadex( 0 );
+ return 0;
+}
+
#include "tchar.h"
- std::string convertTChar( LPCTSTR name )
- {
- std::string s;
+std::string convertTChar( LPCTSTR name )
+{
+ std::string s;
#if defined( UNICODE ) || defined( _UNICODE )
- // Yes, this conversion doesn't make sense for two-byte characters
- // but RtAudio is currently written to return an std::string of
- // one-byte chars for the device name.
- for ( unsigned int i=0; i<wcslen( name ); i++ )
- s.push_back( name[i] );
+ // Yes, this conversion doesn't make sense for two-byte characters
+ // but RtAudio is currently written to return an std::string of
+ // one-byte chars for the device name.
+ for ( unsigned int i=0; i<wcslen( name ); i++ )
+ s.push_back( name[i] );
#else
- s.append( std::string( name ) );
+ s.append( std::string( name ) );
#endif
- return s;
- }
+ return s;
+}
- static BOOL CALLBACK deviceQueryCallback( LPGUID lpguid,
- LPCTSTR description,
- LPCTSTR module,
- LPVOID lpContext )
- {
- EnumInfo *info = (EnumInfo *) lpContext;
+static BOOL CALLBACK deviceQueryCallback( LPGUID lpguid,
+ LPCTSTR description,
+ LPCTSTR module,
+ LPVOID lpContext )
+{
+ bool *isInput = (bool *) lpContext;
- HRESULT hr;
- if ( info->isInput == true ) {
- DSCCAPS caps;
- LPDIRECTSOUNDCAPTURE object;
+ HRESULT hr;
+ bool validDevice = false;
+ if ( *isInput == true ) {
+ DSCCAPS caps;
+ LPDIRECTSOUNDCAPTURE object;
- hr = DirectSoundCaptureCreate( lpguid, &object, NULL );
- if ( hr != DS_OK ) return TRUE;
+ hr = DirectSoundCaptureCreate( lpguid, &object, NULL );
+ if ( hr != DS_OK ) return TRUE;
- caps.dwSize = sizeof(caps);
- hr = object->GetCaps( &caps );
- if ( hr == DS_OK ) {
- if ( caps.dwChannels > 0 && caps.dwFormats > 0 )
- info->counter++;
- }
- object->Release();
+ caps.dwSize = sizeof(caps);
+ hr = object->GetCaps( &caps );
+ if ( hr == DS_OK ) {
+ if ( caps.dwChannels > 0 && caps.dwFormats > 0 )
+ validDevice = true;
}
- else {
- DSCAPS caps;
- LPDIRECTSOUND object;
- hr = DirectSoundCreate( lpguid, &object, NULL );
- if ( hr != DS_OK ) return TRUE;
-
- caps.dwSize = sizeof(caps);
- hr = object->GetCaps( &caps );
- if ( hr == DS_OK ) {
- if ( caps.dwFlags & DSCAPS_PRIMARYMONO || caps.dwFlags & DSCAPS_PRIMARYSTEREO )
- info->counter++;
+ object->Release();
+ }
+ else {
+ DSCAPS caps;
+ LPDIRECTSOUND object;
+ hr = DirectSoundCreate( lpguid, &object, NULL );
+ if ( hr != DS_OK ) return TRUE;
+
+ caps.dwSize = sizeof(caps);
+ hr = object->GetCaps( &caps );
+ if ( hr == DS_OK ) {
+ if ( caps.dwFlags & DSCAPS_PRIMARYMONO || caps.dwFlags & DSCAPS_PRIMARYSTEREO )
+ validDevice = true;
+ }
+ object->Release();
+ }
+
+ // If good device, then save its name and guid.
+ std::string name = convertTChar( description );
+ if ( name == "Primary Sound Driver" || name == "Primary Sound Capture Driver" )
+ name = "Default Device";
+ if ( validDevice ) {
+ for ( unsigned int i=0; i<dsDevices.size(); i++ ) {
+ if ( dsDevices[i].name == name ) {
+ dsDevices[i].found = true;
+ if ( *isInput ) {
+ dsDevices[i].id[1] = lpguid;
+ dsDevices[i].validId[1] = true;
+ }
+ else {
+ dsDevices[i].id[0] = lpguid;
+ dsDevices[i].validId[0] = true;
+ }
+ return TRUE;
}
- object->Release();
}
- if ( info->getDefault && lpguid == NULL ) return FALSE;
-
- if ( info->findIndex && info->counter > info->index ) {
- info->id = lpguid;
- info->name = convertTChar( description );
- return FALSE;
+ DsDevice device;
+ device.name = name;
+ device.found = true;
+ if ( *isInput ) {
+ device.id[1] = lpguid;
+ device.validId[1] = true;
}
-
- return TRUE;
+ else {
+ device.id[0] = lpguid;
+ device.validId[0] = true;
+ }
+ dsDevices.push_back( device );
}
- static char* getErrorString( int code )
- {
- switch ( code ) {
+ return TRUE;
+}
- case DSERR_ALLOCATED:
- return "Already allocated";
+static const char* getErrorString( int code )
+{
+ switch ( code ) {
- case DSERR_CONTROLUNAVAIL:
- return "Control unavailable";
+ case DSERR_ALLOCATED:
+ return "Already allocated";
- case DSERR_INVALIDPARAM:
- return "Invalid parameter";
+ case DSERR_CONTROLUNAVAIL:
+ return "Control unavailable";
- case DSERR_INVALIDCALL:
- return "Invalid call";
+ case DSERR_INVALIDPARAM:
+ return "Invalid parameter";
- case DSERR_GENERIC:
- return "Generic error";
+ case DSERR_INVALIDCALL:
+ return "Invalid call";
- case DSERR_PRIOLEVELNEEDED:
- return "Priority level needed";
+ case DSERR_GENERIC:
+ return "Generic error";
- case DSERR_OUTOFMEMORY:
- return "Out of memory";
+ case DSERR_PRIOLEVELNEEDED:
+ return "Priority level needed";
- case DSERR_BADFORMAT:
- return "The sample rate or the channel format is not supported";
+ case DSERR_OUTOFMEMORY:
+ return "Out of memory";
- case DSERR_UNSUPPORTED:
- return "Not supported";
+ case DSERR_BADFORMAT:
+ return "The sample rate or the channel format is not supported";
- case DSERR_NODRIVER:
- return "No driver";
+ case DSERR_UNSUPPORTED:
+ return "Not supported";
- case DSERR_ALREADYINITIALIZED:
- return "Already initialized";
+ case DSERR_NODRIVER:
+ return "No driver";
- case DSERR_NOAGGREGATION:
- return "No aggregation";
+ case DSERR_ALREADYINITIALIZED:
+ return "Already initialized";
- case DSERR_BUFFERLOST:
- return "Buffer lost";
+ case DSERR_NOAGGREGATION:
+ return "No aggregation";
- case DSERR_OTHERAPPHASPRIO:
- return "Another application already has priority";
+ case DSERR_BUFFERLOST:
+ return "Buffer lost";
- case DSERR_UNINITIALIZED:
- return "Uninitialized";
+ case DSERR_OTHERAPPHASPRIO:
+ return "Another application already has priority";
- default:
- return "DirectSound unknown error";
- }
+ case DSERR_UNINITIALIZED:
+ return "Uninitialized";
+
+ default:
+ return "DirectSound unknown error";
}
- //******************** End of __WINDOWS_DS__ *********************//
+}
+//******************** End of __WINDOWS_DS__ *********************//
#endif
// A structure to hold various information related to the ALSA API
// implementation.
- struct AlsaHandle {
- snd_pcm_t *handles[2];
- bool synchronized;
- bool xrun[2];
- pthread_cond_t runnable;
-
- AlsaHandle()
- :synchronized(false) { xrun[0] = false; xrun[1] = false; }
- };
+struct AlsaHandle {
+ snd_pcm_t *handles[2];
+ bool synchronized;
+ bool xrun[2];
+ pthread_cond_t runnable;
- extern "C" void *alsaCallbackHandler( void * ptr );
+ AlsaHandle()
+ :synchronized(false) { xrun[0] = false; xrun[1] = false; }
+};
- RtApiAlsa :: RtApiAlsa()
- {
- // Nothing to do here.
- }
+extern "C" void *alsaCallbackHandler( void * ptr );
- RtApiAlsa :: ~RtApiAlsa()
- {
- if ( stream_.state != STREAM_CLOSED ) closeStream();
- }
+RtApiAlsa :: RtApiAlsa()
+{
+ // Nothing to do here.
+}
- unsigned int RtApiAlsa :: getDeviceCount( void )
- {
- unsigned nDevices = 0;
- int result, subdevice, card;
- char name[64];
- snd_ctl_t *handle;
+RtApiAlsa :: ~RtApiAlsa()
+{
+ if ( stream_.state != STREAM_CLOSED ) closeStream();
+}
- // Count cards and devices
- card = -1;
- snd_card_next( &card );
- while ( card >= 0 ) {
- sprintf( name, "hw:%d", card );
- result = snd_ctl_open( &handle, name, 0 );
+unsigned int RtApiAlsa :: getDeviceCount( void )
+{
+ unsigned nDevices = 0;
+ int result, subdevice, card;
+ char name[64];
+ snd_ctl_t *handle;
+
+ // Count cards and devices
+ card = -1;
+ snd_card_next( &card );
+ while ( card >= 0 ) {
+ sprintf( name, "hw:%d", card );
+ result = snd_ctl_open( &handle, name, 0 );
+ if ( result < 0 ) {
+ errorStream_ << "RtApiAlsa::getDeviceCount: control open, card = " << card << ", " << snd_strerror( result ) << ".";
+ errorText_ = errorStream_.str();
+ error( RtError::WARNING );
+ goto nextcard;
+ }
+ subdevice = -1;
+ while( 1 ) {
+ result = snd_ctl_pcm_next_device( handle, &subdevice );
if ( result < 0 ) {
- errorStream_ << "RtApiAlsa::getDeviceCount: control open, card = " << card << ", " << snd_strerror( result ) << ".";
+ errorStream_ << "RtApiAlsa::getDeviceCount: control next device, card = " << card << ", " << snd_strerror( result ) << ".";
errorText_ = errorStream_.str();
error( RtError::WARNING );
- goto nextcard;
- }
- subdevice = -1;
- while( 1 ) {
- result = snd_ctl_pcm_next_device( handle, &subdevice );
- if ( result < 0 ) {
- errorStream_ << "RtApiAlsa::getDeviceCount: control next device, card = " << card << ", " << snd_strerror( result ) << ".";
- errorText_ = errorStream_.str();
- error( RtError::WARNING );
- break;
- }
- if ( subdevice < 0 )
- break;
- nDevices++;
+ break;
}
- nextcard:
- snd_ctl_close( handle );
- snd_card_next( &card );
+ if ( subdevice < 0 )
+ break;
+ nDevices++;
}
-
- return nDevices;
+ nextcard:
+ snd_ctl_close( handle );
+ snd_card_next( &card );
}
- RtAudio::DeviceInfo RtApiAlsa :: getDeviceInfo( unsigned int device )
- {
- RtAudio::DeviceInfo info;
- info.probed = false;
+ return nDevices;
+}
- unsigned nDevices = 0;
- int result, subdevice, card;
- char name[64];
- snd_ctl_t *chandle;
+RtAudio::DeviceInfo RtApiAlsa :: getDeviceInfo( unsigned int device )
+{
+ RtAudio::DeviceInfo info;
+ info.probed = false;
- // Count cards and devices
- card = -1;
- snd_card_next( &card );
- while ( card >= 0 ) {
- sprintf( name, "hw:%d", card );
- result = snd_ctl_open( &chandle, name, SND_CTL_NONBLOCK );
+ unsigned nDevices = 0;
+ int result, subdevice, card;
+ char name[64];
+ snd_ctl_t *chandle;
+
+ // Count cards and devices
+ card = -1;
+ snd_card_next( &card );
+ while ( card >= 0 ) {
+ sprintf( name, "hw:%d", card );
+ result = snd_ctl_open( &chandle, name, SND_CTL_NONBLOCK );
+ if ( result < 0 ) {
+ errorStream_ << "RtApiAlsa::getDeviceInfo: control open, card = " << card << ", " << snd_strerror( result ) << ".";
+ errorText_ = errorStream_.str();
+ error( RtError::WARNING );
+ goto nextcard;
+ }
+ subdevice = -1;
+ while( 1 ) {
+ result = snd_ctl_pcm_next_device( chandle, &subdevice );
if ( result < 0 ) {
- errorStream_ << "RtApiAlsa::getDeviceInfo: control open, card = " << card << ", " << snd_strerror( result ) << ".";
+ errorStream_ << "RtApiAlsa::getDeviceInfo: control next device, card = " << card << ", " << snd_strerror( result ) << ".";
errorText_ = errorStream_.str();
error( RtError::WARNING );
- goto nextcard;
+ break;
}
- subdevice = -1;
- while( 1 ) {
- result = snd_ctl_pcm_next_device( chandle, &subdevice );
- if ( result < 0 ) {
- errorStream_ << "RtApiAlsa::getDeviceInfo: control next device, card = " << card << ", " << snd_strerror( result ) << ".";
- errorText_ = errorStream_.str();
- error( RtError::WARNING );
- break;
- }
- if ( subdevice < 0 ) break;
- if ( nDevices == device ) {
- sprintf( name, "hw:%d,%d", card, subdevice );
- goto foundDevice;
- }
- nDevices++;
+ if ( subdevice < 0 ) break;
+ if ( nDevices == device ) {
+ sprintf( name, "hw:%d,%d", card, subdevice );
+ goto foundDevice;
}
- nextcard:
- snd_ctl_close( chandle );
- snd_card_next( &card );
+ nDevices++;
}
+ nextcard:
+ snd_ctl_close( chandle );
+ snd_card_next( &card );
+ }
- if ( nDevices == 0 ) {
- errorText_ = "RtApiAlsa::getDeviceInfo: no devices found!";
- error( RtError::INVALID_USE );
- }
+ if ( nDevices == 0 ) {
+ errorText_ = "RtApiAlsa::getDeviceInfo: no devices found!";
+ error( RtError::INVALID_USE );
+ }
- if ( device >= nDevices ) {
- errorText_ = "RtApiAlsa::getDeviceInfo: device ID is invalid!";
- error( RtError::INVALID_USE );
- }
+ if ( device >= nDevices ) {
+ errorText_ = "RtApiAlsa::getDeviceInfo: device ID is invalid!";
+ error( RtError::INVALID_USE );
+ }
- foundDevice:
+ foundDevice:
- // If a stream is already open, we cannot probe the stream devices.
- // Thus, use the saved results.
- if ( stream_.state != STREAM_CLOSED &&
- ( stream_.device[0] == device || stream_.device[1] == device ) ) {
- if ( device >= devices_.size() ) {
- errorText_ = "RtApiAlsa::getDeviceInfo: device ID was not present before stream was opened.";
- error( RtError::WARNING );
- return info;
- }
- return devices_[ device ];
+ // If a stream is already open, we cannot probe the stream devices.
+ // Thus, use the saved results.
+ if ( stream_.state != STREAM_CLOSED &&
+ ( stream_.device[0] == device || stream_.device[1] == device ) ) {
+ if ( device >= devices_.size() ) {
+ errorText_ = "RtApiAlsa::getDeviceInfo: device ID was not present before stream was opened.";
+ error( RtError::WARNING );
+ return info;
}
+ return devices_[ device ];
+ }
- int openMode = SND_PCM_ASYNC;
- snd_pcm_stream_t stream;
- snd_pcm_info_t *pcminfo;
- snd_pcm_info_alloca( &pcminfo );
- snd_pcm_t *phandle;
- snd_pcm_hw_params_t *params;
- snd_pcm_hw_params_alloca( ¶ms );
+ int openMode = SND_PCM_ASYNC;
+ snd_pcm_stream_t stream;
+ snd_pcm_info_t *pcminfo;
+ snd_pcm_info_alloca( &pcminfo );
+ snd_pcm_t *phandle;
+ snd_pcm_hw_params_t *params;
+ snd_pcm_hw_params_alloca( ¶ms );
- // First try for playback
- stream = SND_PCM_STREAM_PLAYBACK;
- snd_pcm_info_set_device( pcminfo, subdevice );
- snd_pcm_info_set_subdevice( pcminfo, 0 );
- snd_pcm_info_set_stream( pcminfo, stream );
+ // First try for playback
+ stream = SND_PCM_STREAM_PLAYBACK;
+ snd_pcm_info_set_device( pcminfo, subdevice );
+ snd_pcm_info_set_subdevice( pcminfo, 0 );
+ snd_pcm_info_set_stream( pcminfo, stream );
- result = snd_ctl_pcm_info( chandle, pcminfo );
- if ( result < 0 ) {
- // Device probably doesn't support playback.
- goto captureProbe;
- }
-
- result = snd_pcm_open( &phandle, name, stream, openMode | SND_PCM_NONBLOCK );
- if ( result < 0 ) {
- errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_open error for device (" << name << "), " << snd_strerror( result ) << ".";
- errorText_ = errorStream_.str();
- error( RtError::WARNING );
- goto captureProbe;
- }
+ result = snd_ctl_pcm_info( chandle, pcminfo );
+ if ( result < 0 ) {
+ // Device probably doesn't support playback.
+ goto captureProbe;
+ }
- // The device is open ... fill the parameter structure.
- result = snd_pcm_hw_params_any( phandle, params );
- if ( result < 0 ) {
- snd_pcm_close( phandle );
- errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_hw_params error for device (" << name << "), " << snd_strerror( result ) << ".";
- errorText_ = errorStream_.str();
- error( RtError::WARNING );
- goto captureProbe;
- }
+ result = snd_pcm_open( &phandle, name, stream, openMode | SND_PCM_NONBLOCK );
+ if ( result < 0 ) {
+ errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_open error for device (" << name << "), " << snd_strerror( result ) << ".";
+ errorText_ = errorStream_.str();
+ error( RtError::WARNING );
+ goto captureProbe;
+ }
- // Get output channel information.
- unsigned int value;
- result = snd_pcm_hw_params_get_channels_max( params, &value );
- if ( result < 0 ) {
- snd_pcm_close( phandle );
- errorStream_ << "RtApiAlsa::getDeviceInfo: error getting device (" << name << ") output channels, " << snd_strerror( result ) << ".";
- errorText_ = errorStream_.str();
- error( RtError::WARNING );
- goto captureProbe;
- }
- info.outputChannels = value;
+ // The device is open ... fill the parameter structure.
+ result = snd_pcm_hw_params_any( phandle, params );
+ if ( result < 0 ) {
snd_pcm_close( phandle );
+ errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_hw_params error for device (" << name << "), " << snd_strerror( result ) << ".";
+ errorText_ = errorStream_.str();
+ error( RtError::WARNING );
+ goto captureProbe;
+ }
- captureProbe:
- // Now try for capture
- stream = SND_PCM_STREAM_CAPTURE;
- snd_pcm_info_set_stream( pcminfo, stream );
+ // Get output channel information.
+ unsigned int value;
+ result = snd_pcm_hw_params_get_channels_max( params, &value );
+ if ( result < 0 ) {
+ snd_pcm_close( phandle );
+ errorStream_ << "RtApiAlsa::getDeviceInfo: error getting device (" << name << ") output channels, " << snd_strerror( result ) << ".";
+ errorText_ = errorStream_.str();
+ error( RtError::WARNING );
+ goto captureProbe;
+ }
+ info.outputChannels = value;
+ snd_pcm_close( phandle );
- result = snd_ctl_pcm_info( chandle, pcminfo );
- snd_ctl_close( chandle );
- if ( result < 0 ) {
- // Device probably doesn't support capture.
- if ( info.outputChannels == 0 ) return info;
- goto probeParameters;
- }
+ captureProbe:
+ // Now try for capture
+ stream = SND_PCM_STREAM_CAPTURE;
+ snd_pcm_info_set_stream( pcminfo, stream );
- result = snd_pcm_open( &phandle, name, stream, openMode | SND_PCM_NONBLOCK);
- if ( result < 0 ) {
- errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_open error for device (" << name << "), " << snd_strerror( result ) << ".";
- errorText_ = errorStream_.str();
- error( RtError::WARNING );
- if ( info.outputChannels == 0 ) return info;
- goto probeParameters;
- }
+ result = snd_ctl_pcm_info( chandle, pcminfo );
+ snd_ctl_close( chandle );
+ if ( result < 0 ) {
+ // Device probably doesn't support capture.
+ if ( info.outputChannels == 0 ) return info;
+ goto probeParameters;
+ }
- // The device is open ... fill the parameter structure.
- result = snd_pcm_hw_params_any( phandle, params );
- if ( result < 0 ) {
- snd_pcm_close( phandle );
- errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_hw_params error for device (" << name << "), " << snd_strerror( result ) << ".";
- errorText_ = errorStream_.str();
- error( RtError::WARNING );
- if ( info.outputChannels == 0 ) return info;
- goto probeParameters;
- }
+ result = snd_pcm_open( &phandle, name, stream, openMode | SND_PCM_NONBLOCK);
+ if ( result < 0 ) {
+ errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_open error for device (" << name << "), " << snd_strerror( result ) << ".";
+ errorText_ = errorStream_.str();
+ error( RtError::WARNING );
+ if ( info.outputChannels == 0 ) return info;
+ goto probeParameters;
+ }
- result = snd_pcm_hw_params_get_channels_max( params, &value );
- if ( result < 0 ) {
- snd_pcm_close( phandle );
- errorStream_ << "RtApiAlsa::getDeviceInfo: error getting device (" << name << ") input channels, " << snd_strerror( result ) << ".";
- errorText_ = errorStream_.str();
- error( RtError::WARNING );
- if ( info.outputChannels == 0 ) return info;
- goto probeParameters;
- }
- info.inputChannels = value;
+ // The device is open ... fill the parameter structure.
+ result = snd_pcm_hw_params_any( phandle, params );
+ if ( result < 0 ) {
snd_pcm_close( phandle );
+ errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_hw_params error for device (" << name << "), " << snd_strerror( result ) << ".";
+ errorText_ = errorStream_.str();
+ error( RtError::WARNING );
+ if ( info.outputChannels == 0 ) return info;
+ goto probeParameters;
+ }
- // If device opens for both playback and capture, we determine the channels.
- if ( info.outputChannels > 0 && info.inputChannels > 0 )
- info.duplexChannels = (info.outputChannels > info.inputChannels) ? info.inputChannels : info.outputChannels;
+ result = snd_pcm_hw_params_get_channels_max( params, &value );
+ if ( result < 0 ) {
+ snd_pcm_close( phandle );
+ errorStream_ << "RtApiAlsa::getDeviceInfo: error getting device (" << name << ") input channels, " << snd_strerror( result ) << ".";
+ errorText_ = errorStream_.str();
+ error( RtError::WARNING );
+ if ( info.outputChannels == 0 ) return info;
+ goto probeParameters;
+ }
+ info.inputChannels = value;
+ snd_pcm_close( phandle );
- // ALSA doesn't provide default devices so we'll use the first available one.
- if ( device == 0 && info.outputChannels > 0 )
- info.isDefaultOutput = true;
- if ( device == 0 && info.inputChannels > 0 )
- info.isDefaultInput = true;
-
- probeParameters:
- // At this point, we just need to figure out the supported data
- // formats and sample rates. We'll proceed by opening the device in
- // the direction with the maximum number of channels, or playback if
- // they are equal. This might limit our sample rate options, but so
- // be it.
-
- if ( info.outputChannels >= info.inputChannels )
- stream = SND_PCM_STREAM_PLAYBACK;
- else
- stream = SND_PCM_STREAM_CAPTURE;
- snd_pcm_info_set_stream( pcminfo, stream );
+ // If device opens for both playback and capture, we determine the channels.
+ if ( info.outputChannels > 0 && info.inputChannels > 0 )
+ info.duplexChannels = (info.outputChannels > info.inputChannels) ? info.inputChannels : info.outputChannels;
- result = snd_pcm_open( &phandle, name, stream, openMode | SND_PCM_NONBLOCK);
- if ( result < 0 ) {
- errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_open error for device (" << name << "), " << snd_strerror( result ) << ".";
- errorText_ = errorStream_.str();
- error( RtError::WARNING );
- return info;
- }
+ // ALSA doesn't provide default devices so we'll use the first available one.
+ if ( device == 0 && info.outputChannels > 0 )
+ info.isDefaultOutput = true;
+ if ( device == 0 && info.inputChannels > 0 )
+ info.isDefaultInput = true;
- // The device is open ... fill the parameter structure.
- result = snd_pcm_hw_params_any( phandle, params );
- if ( result < 0 ) {
- snd_pcm_close( phandle );
- errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_hw_params error for device (" << name << "), " << snd_strerror( result ) << ".";
- errorText_ = errorStream_.str();
- error( RtError::WARNING );
- return info;
- }
+ probeParameters:
+ // At this point, we just need to figure out the supported data
+ // formats and sample rates. We'll proceed by opening the device in
+ // the direction with the maximum number of channels, or playback if
+ // they are equal. This might limit our sample rate options, but so
+ // be it.
- // Test our discrete set of sample rate values.
- info.sampleRates.clear();
- for ( unsigned int i=0; i<MAX_SAMPLE_RATES; i++ ) {
- if ( snd_pcm_hw_params_test_rate( phandle, params, SAMPLE_RATES[i], 0 ) == 0 )
- info.sampleRates.push_back( SAMPLE_RATES[i] );
- }
- if ( info.sampleRates.size() == 0 ) {
- snd_pcm_close( phandle );
- errorStream_ << "RtApiAlsa::getDeviceInfo: no supported sample rates found for device (" << name << ").";
- errorText_ = errorStream_.str();
- error( RtError::WARNING );
- return info;
- }
-
- // Probe the supported data formats ... we don't care about endian-ness just yet
- snd_pcm_format_t format;
- info.nativeFormats = 0;
- format = SND_PCM_FORMAT_S8;
- if ( snd_pcm_hw_params_test_format( phandle, params, format ) == 0 )
- info.nativeFormats |= RTAUDIO_SINT8;
- format = SND_PCM_FORMAT_S16;
- if ( snd_pcm_hw_params_test_format( phandle, params, format ) == 0 )
- info.nativeFormats |= RTAUDIO_SINT16;
- format = SND_PCM_FORMAT_S24;
- if ( snd_pcm_hw_params_test_format( phandle, params, format ) == 0 )
- info.nativeFormats |= RTAUDIO_SINT24;
- format = SND_PCM_FORMAT_S32;
- if ( snd_pcm_hw_params_test_format( phandle, params, format ) == 0 )
- info.nativeFormats |= RTAUDIO_SINT32;
- format = SND_PCM_FORMAT_FLOAT;
- if ( snd_pcm_hw_params_test_format( phandle, params, format ) == 0 )
- info.nativeFormats |= RTAUDIO_FLOAT32;
- format = SND_PCM_FORMAT_FLOAT64;
- if ( snd_pcm_hw_params_test_format( phandle, params, format ) == 0 )
- info.nativeFormats |= RTAUDIO_FLOAT64;
-
- // Check that we have at least one supported format
- if ( info.nativeFormats == 0 ) {
- errorStream_ << "RtApiAlsa::getDeviceInfo: pcm device (" << name << ") data format not supported by RtAudio.";
- errorText_ = errorStream_.str();
- error( RtError::WARNING );
- return info;
- }
+ if ( info.outputChannels >= info.inputChannels )
+ stream = SND_PCM_STREAM_PLAYBACK;
+ else
+ stream = SND_PCM_STREAM_CAPTURE;
+ snd_pcm_info_set_stream( pcminfo, stream );
- // Get the device name
- char *cardname;
- result = snd_card_get_name( card, &cardname );
- if ( result >= 0 )
- sprintf( name, "hw:%s,%d", cardname, subdevice );
- info.name = name;
+ result = snd_pcm_open( &phandle, name, stream, openMode | SND_PCM_NONBLOCK);
+ if ( result < 0 ) {
+ errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_open error for device (" << name << "), " << snd_strerror( result ) << ".";
+ errorText_ = errorStream_.str();
+ error( RtError::WARNING );
+ return info;
+ }
- // That's all ... close the device and return
+ // The device is open ... fill the parameter structure.
+ result = snd_pcm_hw_params_any( phandle, params );
+ if ( result < 0 ) {
snd_pcm_close( phandle );
- info.probed = true;
+ errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_hw_params error for device (" << name << "), " << snd_strerror( result ) << ".";
+ errorText_ = errorStream_.str();
+ error( RtError::WARNING );
return info;
}
- void RtApiAlsa :: saveDeviceInfo( void )
- {
- devices_.clear();
+ // Test our discrete set of sample rate values.
+ info.sampleRates.clear();
+ for ( unsigned int i=0; i<MAX_SAMPLE_RATES; i++ ) {
+ if ( snd_pcm_hw_params_test_rate( phandle, params, SAMPLE_RATES[i], 0 ) == 0 )
+ info.sampleRates.push_back( SAMPLE_RATES[i] );
+ }
+ if ( info.sampleRates.size() == 0 ) {
+ snd_pcm_close( phandle );
+ errorStream_ << "RtApiAlsa::getDeviceInfo: no supported sample rates found for device (" << name << ").";
+ errorText_ = errorStream_.str();
+ error( RtError::WARNING );
+ return info;
+ }
- unsigned int nDevices = getDeviceCount();
- devices_.resize( nDevices );
- for ( unsigned int i=0; i<nDevices; i++ )
- devices_[i] = getDeviceInfo( i );
+ // Probe the supported data formats ... we don't care about endian-ness just yet
+ snd_pcm_format_t format;
+ info.nativeFormats = 0;
+ format = SND_PCM_FORMAT_S8;
+ if ( snd_pcm_hw_params_test_format( phandle, params, format ) == 0 )
+ info.nativeFormats |= RTAUDIO_SINT8;
+ format = SND_PCM_FORMAT_S16;
+ if ( snd_pcm_hw_params_test_format( phandle, params, format ) == 0 )
+ info.nativeFormats |= RTAUDIO_SINT16;
+ format = SND_PCM_FORMAT_S24;
+ if ( snd_pcm_hw_params_test_format( phandle, params, format ) == 0 )
+ info.nativeFormats |= RTAUDIO_SINT24;
+ format = SND_PCM_FORMAT_S32;
+ if ( snd_pcm_hw_params_test_format( phandle, params, format ) == 0 )
+ info.nativeFormats |= RTAUDIO_SINT32;
+ format = SND_PCM_FORMAT_FLOAT;
+ if ( snd_pcm_hw_params_test_format( phandle, params, format ) == 0 )
+ info.nativeFormats |= RTAUDIO_FLOAT32;
+ format = SND_PCM_FORMAT_FLOAT64;
+ if ( snd_pcm_hw_params_test_format( phandle, params, format ) == 0 )
+ info.nativeFormats |= RTAUDIO_FLOAT64;
+
+ // Check that we have at least one supported format
+ if ( info.nativeFormats == 0 ) {
+ errorStream_ << "RtApiAlsa::getDeviceInfo: pcm device (" << name << ") data format not supported by RtAudio.";
+ errorText_ = errorStream_.str();
+ error( RtError::WARNING );
+ return info;
}
- bool RtApiAlsa :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,
- unsigned int firstChannel, unsigned int sampleRate,
- RtAudioFormat format, unsigned int *bufferSize,
- RtAudio::StreamOptions *options )
+ // Get the device name
+ char *cardname;
+ result = snd_card_get_name( card, &cardname );
+ if ( result >= 0 )
+ sprintf( name, "hw:%s,%d", cardname, subdevice );
+ info.name = name;
- {
+ // That's all ... close the device and return
+ snd_pcm_close( phandle );
+ info.probed = true;
+ return info;
+}
+
+void RtApiAlsa :: saveDeviceInfo( void )
+{
+ devices_.clear();
+
+ unsigned int nDevices = getDeviceCount();
+ devices_.resize( nDevices );
+ for ( unsigned int i=0; i<nDevices; i++ )
+ devices_[i] = getDeviceInfo( i );
+}
+
+bool RtApiAlsa :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,
+ unsigned int firstChannel, unsigned int sampleRate,
+ RtAudioFormat format, unsigned int *bufferSize,
+ RtAudio::StreamOptions *options )
+
+{
#if defined(__RTAUDIO_DEBUG__)
- snd_output_t *out;
- snd_output_stdio_attach(&out, stderr, 0);
+ snd_output_t *out;
+ snd_output_stdio_attach(&out, stderr, 0);
#endif
- // I'm not using the "plug" interface ... too much inconsistent behavior.
+ // I'm not using the "plug" interface ... too much inconsistent behavior.
- unsigned nDevices = 0;
- int result, subdevice, card;
- char name[64];
- snd_ctl_t *chandle;
+ unsigned nDevices = 0;
+ int result, subdevice, card;
+ char name[64];
+ snd_ctl_t *chandle;
- // Count cards and devices
- card = -1;
- snd_card_next( &card );
- while ( card >= 0 ) {
- sprintf( name, "hw:%d", card );
- result = snd_ctl_open( &chandle, name, SND_CTL_NONBLOCK );
- if ( result < 0 ) {
- errorStream_ << "RtApiAlsa::probeDeviceOpen: control open, card = " << card << ", " << snd_strerror( result ) << ".";
- errorText_ = errorStream_.str();
- return FAILURE;
- }
- subdevice = -1;
- while( 1 ) {
- result = snd_ctl_pcm_next_device( chandle, &subdevice );
- if ( result < 0 ) break;
- if ( subdevice < 0 ) break;
- if ( nDevices == device ) {
- sprintf( name, "hw:%d,%d", card, subdevice );
- snd_ctl_close( chandle );
- goto foundDevice;
- }
- nDevices++;
+ // Count cards and devices
+ card = -1;
+ snd_card_next( &card );
+ while ( card >= 0 ) {
+ sprintf( name, "hw:%d", card );
+ result = snd_ctl_open( &chandle, name, SND_CTL_NONBLOCK );
+ if ( result < 0 ) {
+ errorStream_ << "RtApiAlsa::probeDeviceOpen: control open, card = " << card << ", " << snd_strerror( result ) << ".";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
+ subdevice = -1;
+ while( 1 ) {
+ result = snd_ctl_pcm_next_device( chandle, &subdevice );
+ if ( result < 0 ) break;
+ if ( subdevice < 0 ) break;
+ if ( nDevices == device ) {
+ sprintf( name, "hw:%d,%d", card, subdevice );
+ snd_ctl_close( chandle );
+ goto foundDevice;
}
- snd_ctl_close( chandle );
- snd_card_next( &card );
+ nDevices++;
}
+ snd_ctl_close( chandle );
+ snd_card_next( &card );
+ }
- if ( nDevices == 0 ) {
- // This should not happen because a check is made before this function is called.
- errorText_ = "RtApiAlsa::probeDeviceOpen: no devices found!";
- return FAILURE;
- }
+ if ( nDevices == 0 ) {
+ // This should not happen because a check is made before this function is called.
+ errorText_ = "RtApiAlsa::probeDeviceOpen: no devices found!";
+ return FAILURE;
+ }
- if ( device >= nDevices ) {
- // This should not happen because a check is made before this function is called.
- errorText_ = "RtApiAlsa::probeDeviceOpen: device ID is invalid!";
- return FAILURE;
- }
+ if ( device >= nDevices ) {
+ // This should not happen because a check is made before this function is called.
+ errorText_ = "RtApiAlsa::probeDeviceOpen: device ID is invalid!";
+ return FAILURE;
+ }
- foundDevice:
+ foundDevice:
- // The getDeviceInfo() function will not work for a device that is
- // already open. Thus, we'll probe the system before opening a
- // stream and save the results for use by getDeviceInfo().
- if ( mode == OUTPUT || ( mode == INPUT && stream_.mode != OUTPUT ) ) // only do once
- this->saveDeviceInfo();
+ // The getDeviceInfo() function will not work for a device that is
+ // already open. Thus, we'll probe the system before opening a
+ // stream and save the results for use by getDeviceInfo().
+ if ( mode == OUTPUT || ( mode == INPUT && stream_.mode != OUTPUT ) ) // only do once
+ this->saveDeviceInfo();
- snd_pcm_stream_t stream;
+ snd_pcm_stream_t stream;
+ if ( mode == OUTPUT )
+ stream = SND_PCM_STREAM_PLAYBACK;
+ else
+ stream = SND_PCM_STREAM_CAPTURE;
+
+ snd_pcm_t *phandle;
+ int openMode = SND_PCM_ASYNC;
+ result = snd_pcm_open( &phandle, name, stream, openMode );
+ if ( result < 0 ) {
if ( mode == OUTPUT )
- stream = SND_PCM_STREAM_PLAYBACK;
+ errorStream_ << "RtApiAlsa::probeDeviceOpen: pcm device (" << name << ") won't open for output.";
else
- stream = SND_PCM_STREAM_CAPTURE;
-
- snd_pcm_t *phandle;
- int openMode = SND_PCM_ASYNC;
- result = snd_pcm_open( &phandle, name, stream, openMode );
- if ( result < 0 ) {
- if ( mode == OUTPUT )
- errorStream_ << "RtApiAlsa::probeDeviceOpen: pcm device (" << name << ") won't open for output.";
- else
- errorStream_ << "RtApiAlsa::probeDeviceOpen: pcm device (" << name << ") won't open for input.";
- errorText_ = errorStream_.str();
- return FAILURE;
- }
+ errorStream_ << "RtApiAlsa::probeDeviceOpen: pcm device (" << name << ") won't open for input.";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
- // Fill the parameter structure.
- snd_pcm_hw_params_t *hw_params;
- snd_pcm_hw_params_alloca( &hw_params );
- result = snd_pcm_hw_params_any( phandle, hw_params );
- if ( result < 0 ) {
- snd_pcm_close( phandle );
- errorStream_ << "RtApiAlsa::probeDeviceOpen: error getting pcm device (" << name << ") parameters, " << snd_strerror( result ) << ".";
- errorText_ = errorStream_.str();
- return FAILURE;
- }
+ // Fill the parameter structure.
+ snd_pcm_hw_params_t *hw_params;
+ snd_pcm_hw_params_alloca( &hw_params );
+ result = snd_pcm_hw_params_any( phandle, hw_params );
+ if ( result < 0 ) {
+ snd_pcm_close( phandle );
+ errorStream_ << "RtApiAlsa::probeDeviceOpen: error getting pcm device (" << name << ") parameters, " << snd_strerror( result ) << ".";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
#if defined(__RTAUDIO_DEBUG__)
- fprintf( stderr, "\nRtApiAlsa: dump hardware params just after device open:\n\n" );
- snd_pcm_hw_params_dump( hw_params, out );
+ fprintf( stderr, "\nRtApiAlsa: dump hardware params just after device open:\n\n" );
+ snd_pcm_hw_params_dump( hw_params, out );
#endif
- // Set access ... check user preference.
- if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) {
- stream_.userInterleaved = false;
- result = snd_pcm_hw_params_set_access( phandle, hw_params, SND_PCM_ACCESS_RW_NONINTERLEAVED );
- if ( result < 0 ) {
- result = snd_pcm_hw_params_set_access( phandle, hw_params, SND_PCM_ACCESS_RW_INTERLEAVED );
- stream_.deviceInterleaved[mode] = true;
- }
- else
- stream_.deviceInterleaved[mode] = false;
- }
- else {
- stream_.userInterleaved = true;
+ // Set access ... check user preference.
+ if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) {
+ stream_.userInterleaved = false;
+ result = snd_pcm_hw_params_set_access( phandle, hw_params, SND_PCM_ACCESS_RW_NONINTERLEAVED );
+ if ( result < 0 ) {
result = snd_pcm_hw_params_set_access( phandle, hw_params, SND_PCM_ACCESS_RW_INTERLEAVED );
- if ( result < 0 ) {
- result = snd_pcm_hw_params_set_access( phandle, hw_params, SND_PCM_ACCESS_RW_NONINTERLEAVED );
- stream_.deviceInterleaved[mode] = false;
- }
- else
- stream_.deviceInterleaved[mode] = true;
+ stream_.deviceInterleaved[mode] = true;
}
-
+ else
+ stream_.deviceInterleaved[mode] = false;
+ }
+ else {
+ stream_.userInterleaved = true;
+ result = snd_pcm_hw_params_set_access( phandle, hw_params, SND_PCM_ACCESS_RW_INTERLEAVED );
if ( result < 0 ) {
- snd_pcm_close( phandle );
- errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting pcm device (" << name << ") access, " << snd_strerror( result ) << ".";
- errorText_ = errorStream_.str();
- return FAILURE;
+ result = snd_pcm_hw_params_set_access( phandle, hw_params, SND_PCM_ACCESS_RW_NONINTERLEAVED );
+ stream_.deviceInterleaved[mode] = false;
}
+ else
+ stream_.deviceInterleaved[mode] = true;
+ }
- // Determine how to set the device format.
- stream_.userFormat = format;
- snd_pcm_format_t deviceFormat = SND_PCM_FORMAT_UNKNOWN;
-
- if ( format == RTAUDIO_SINT8 )
- deviceFormat = SND_PCM_FORMAT_S8;
- else if ( format == RTAUDIO_SINT16 )
- deviceFormat = SND_PCM_FORMAT_S16;
- else if ( format == RTAUDIO_SINT24 )
- deviceFormat = SND_PCM_FORMAT_S24;
- else if ( format == RTAUDIO_SINT32 )
- deviceFormat = SND_PCM_FORMAT_S32;
- else if ( format == RTAUDIO_FLOAT32 )
- deviceFormat = SND_PCM_FORMAT_FLOAT;
- else if ( format == RTAUDIO_FLOAT64 )
- deviceFormat = SND_PCM_FORMAT_FLOAT64;
-
- if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat) == 0) {
- stream_.deviceFormat[mode] = format;
- goto setFormat;
- }
+ if ( result < 0 ) {
+ snd_pcm_close( phandle );
+ errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting pcm device (" << name << ") access, " << snd_strerror( result ) << ".";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
- // The user requested format is not natively supported by the device.
- deviceFormat = SND_PCM_FORMAT_FLOAT64;
- if ( snd_pcm_hw_params_test_format( phandle, hw_params, deviceFormat ) == 0 ) {
- stream_.deviceFormat[mode] = RTAUDIO_FLOAT64;
- goto setFormat;
- }
+ // Determine how to set the device format.
+ stream_.userFormat = format;
+ snd_pcm_format_t deviceFormat = SND_PCM_FORMAT_UNKNOWN;
+ if ( format == RTAUDIO_SINT8 )
+ deviceFormat = SND_PCM_FORMAT_S8;
+ else if ( format == RTAUDIO_SINT16 )
+ deviceFormat = SND_PCM_FORMAT_S16;
+ else if ( format == RTAUDIO_SINT24 )
+ deviceFormat = SND_PCM_FORMAT_S24;
+ else if ( format == RTAUDIO_SINT32 )
+ deviceFormat = SND_PCM_FORMAT_S32;
+ else if ( format == RTAUDIO_FLOAT32 )
deviceFormat = SND_PCM_FORMAT_FLOAT;
- if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat ) == 0 ) {
- stream_.deviceFormat[mode] = RTAUDIO_FLOAT32;
- goto setFormat;
- }
+ else if ( format == RTAUDIO_FLOAT64 )
+ deviceFormat = SND_PCM_FORMAT_FLOAT64;
- deviceFormat = SND_PCM_FORMAT_S32;
- if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat ) == 0 ) {
- stream_.deviceFormat[mode] = RTAUDIO_SINT32;
- goto setFormat;
- }
+ if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat) == 0) {
+ stream_.deviceFormat[mode] = format;
+ goto setFormat;
+ }
- deviceFormat = SND_PCM_FORMAT_S24;
- if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat ) == 0 ) {
- stream_.deviceFormat[mode] = RTAUDIO_SINT24;
- goto setFormat;
- }
+ // The user requested format is not natively supported by the device.
+ deviceFormat = SND_PCM_FORMAT_FLOAT64;
+ if ( snd_pcm_hw_params_test_format( phandle, hw_params, deviceFormat ) == 0 ) {
+ stream_.deviceFormat[mode] = RTAUDIO_FLOAT64;
+ goto setFormat;
+ }
- deviceFormat = SND_PCM_FORMAT_S16;
- if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat ) == 0 ) {
- stream_.deviceFormat[mode] = RTAUDIO_SINT16;
- goto setFormat;
- }
+ deviceFormat = SND_PCM_FORMAT_FLOAT;
+ if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat ) == 0 ) {
+ stream_.deviceFormat[mode] = RTAUDIO_FLOAT32;
+ goto setFormat;
+ }
- deviceFormat = SND_PCM_FORMAT_S8;
- if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat ) == 0 ) {
- stream_.deviceFormat[mode] = RTAUDIO_SINT8;
- goto setFormat;
- }
+ deviceFormat = SND_PCM_FORMAT_S32;
+ if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat ) == 0 ) {
+ stream_.deviceFormat[mode] = RTAUDIO_SINT32;
+ goto setFormat;
+ }
+
+ deviceFormat = SND_PCM_FORMAT_S24;
+ if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat ) == 0 ) {
+ stream_.deviceFormat[mode] = RTAUDIO_SINT24;
+ goto setFormat;
+ }
+
+ deviceFormat = SND_PCM_FORMAT_S16;
+ if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat ) == 0 ) {
+ stream_.deviceFormat[mode] = RTAUDIO_SINT16;
+ goto setFormat;
+ }
- // If we get here, no supported format was found.
- errorStream_ << "RtApiAlsa::probeDeviceOpen: pcm device " << device << " data format not supported by RtAudio.";
+ deviceFormat = SND_PCM_FORMAT_S8;
+ if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat ) == 0 ) {
+ stream_.deviceFormat[mode] = RTAUDIO_SINT8;
+ goto setFormat;
+ }
+
+ // If we get here, no supported format was found.
+ errorStream_ << "RtApiAlsa::probeDeviceOpen: pcm device " << device << " data format not supported by RtAudio.";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+
+ setFormat:
+ result = snd_pcm_hw_params_set_format( phandle, hw_params, deviceFormat );
+ if ( result < 0 ) {
+ snd_pcm_close( phandle );
+ errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting pcm device (" << name << ") data format, " << snd_strerror( result ) << ".";
errorText_ = errorStream_.str();
return FAILURE;
+ }
- setFormat:
- result = snd_pcm_hw_params_set_format( phandle, hw_params, deviceFormat );
- if ( result < 0 ) {
+ // Determine whether byte-swaping is necessary.
+ stream_.doByteSwap[mode] = false;
+ if ( deviceFormat != SND_PCM_FORMAT_S8 ) {
+ result = snd_pcm_format_cpu_endian( deviceFormat );
+ if ( result == 0 )
+ stream_.doByteSwap[mode] = true;
+ else if (result < 0) {
snd_pcm_close( phandle );
- errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting pcm device (" << name << ") data format, " << snd_strerror( result ) << ".";
+ errorStream_ << "RtApiAlsa::probeDeviceOpen: error getting pcm device (" << name << ") endian-ness, " << snd_strerror( result ) << ".";
errorText_ = errorStream_.str();
return FAILURE;
}
+ }
- // Determine whether byte-swaping is necessary.
- stream_.doByteSwap[mode] = false;
- if ( deviceFormat != SND_PCM_FORMAT_S8 ) {
- result = snd_pcm_format_cpu_endian( deviceFormat );
- if ( result == 0 )
- stream_.doByteSwap[mode] = true;
- else if (result < 0) {
- snd_pcm_close( phandle );
- errorStream_ << "RtApiAlsa::probeDeviceOpen: error getting pcm device (" << name << ") endian-ness, " << snd_strerror( result ) << ".";
- errorText_ = errorStream_.str();
- return FAILURE;
- }
- }
-
- // Set the sample rate.
- result = snd_pcm_hw_params_set_rate_near( phandle, hw_params, (unsigned int*) &sampleRate, 0 );
- if ( result < 0 ) {
- snd_pcm_close( phandle );
- errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting sample rate on device (" << name << "), " << snd_strerror( result ) << ".";
- errorText_ = errorStream_.str();
- return FAILURE;
- }
+ // Set the sample rate.
+ result = snd_pcm_hw_params_set_rate_near( phandle, hw_params, (unsigned int*) &sampleRate, 0 );
+ if ( result < 0 ) {
+ snd_pcm_close( phandle );
+ errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting sample rate on device (" << name << "), " << snd_strerror( result ) << ".";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
- // Determine the number of channels for this device. We support a possible
- // minimum device channel number > than the value requested by the user.
- stream_.nUserChannels[mode] = channels;
- unsigned int value;
- result = snd_pcm_hw_params_get_channels_max( hw_params, &value );
- unsigned int deviceChannels = value;
- if ( result < 0 || deviceChannels < channels + firstChannel ) {
- snd_pcm_close( phandle );
- errorStream_ << "RtApiAlsa::probeDeviceOpen: requested channel parameters not supported by device (" << name << "), " << snd_strerror( result ) << ".";
- errorText_ = errorStream_.str();
- return FAILURE;
- }
+ // Determine the number of channels for this device. We support a possible
+ // minimum device channel number > than the value requested by the user.
+ stream_.nUserChannels[mode] = channels;
+ unsigned int value;
+ result = snd_pcm_hw_params_get_channels_max( hw_params, &value );
+ unsigned int deviceChannels = value;
+ if ( result < 0 || deviceChannels < channels + firstChannel ) {
+ snd_pcm_close( phandle );
+ errorStream_ << "RtApiAlsa::probeDeviceOpen: requested channel parameters not supported by device (" << name << "), " << snd_strerror( result ) << ".";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
- result = snd_pcm_hw_params_get_channels_min( hw_params, &value );
- if ( result < 0 ) {
- snd_pcm_close( phandle );
- errorStream_ << "RtApiAlsa::probeDeviceOpen: error getting minimum channels for device (" << name << "), " << snd_strerror( result ) << ".";
- errorText_ = errorStream_.str();
- return FAILURE;
- }
- deviceChannels = value;
- if ( deviceChannels < channels + firstChannel ) deviceChannels = channels + firstChannel;
- stream_.nDeviceChannels[mode] = deviceChannels;
+ result = snd_pcm_hw_params_get_channels_min( hw_params, &value );
+ if ( result < 0 ) {
+ snd_pcm_close( phandle );
+ errorStream_ << "RtApiAlsa::probeDeviceOpen: error getting minimum channels for device (" << name << "), " << snd_strerror( result ) << ".";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
+ deviceChannels = value;
+ if ( deviceChannels < channels + firstChannel ) deviceChannels = channels + firstChannel;
+ stream_.nDeviceChannels[mode] = deviceChannels;
- // Set the device channels.
- result = snd_pcm_hw_params_set_channels( phandle, hw_params, deviceChannels );
- if ( result < 0 ) {
- snd_pcm_close( phandle );
- errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting channels for device (" << name << "), " << snd_strerror( result ) << ".";
- errorText_ = errorStream_.str();
- return FAILURE;
- }
+ // Set the device channels.
+ result = snd_pcm_hw_params_set_channels( phandle, hw_params, deviceChannels );
+ if ( result < 0 ) {
+ snd_pcm_close( phandle );
+ errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting channels for device (" << name << "), " << snd_strerror( result ) << ".";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
- // Set the buffer number, which in ALSA is referred to as the "period".
- int totalSize, dir;
- unsigned int periods = 0;
- if ( options ) periods = options->numberOfBuffers;
- totalSize = *bufferSize * periods;
+ // Set the buffer number, which in ALSA is referred to as the "period".
+ int totalSize, dir = 0;
+ unsigned int periods = 0;
+ if ( options ) periods = options->numberOfBuffers;
+ totalSize = *bufferSize * periods;
- // Set the buffer (or period) size.
- snd_pcm_uframes_t periodSize = *bufferSize;
- result = snd_pcm_hw_params_set_period_size_near( phandle, hw_params, &periodSize, &dir );
- if ( result < 0 ) {
- snd_pcm_close( phandle );
- errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting period size for device (" << name << "), " << snd_strerror( result ) << ".";
- errorText_ = errorStream_.str();
- return FAILURE;
- }
- *bufferSize = periodSize;
+ // Set the buffer (or period) size.
+ snd_pcm_uframes_t periodSize = *bufferSize;
+ result = snd_pcm_hw_params_set_period_size_near( phandle, hw_params, &periodSize, &dir );
+ if ( result < 0 ) {
+ snd_pcm_close( phandle );
+ errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting period size for device (" << name << "), " << snd_strerror( result ) << ".";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
+ *bufferSize = periodSize;
- if ( options && options->flags & RTAUDIO_MINIMIZE_LATENCY ) periods = 2;
- else periods = totalSize / *bufferSize;
- // Even though the hardware might allow 1 buffer, it won't work reliably.
- if ( periods < 2 ) periods = 2;
- result = snd_pcm_hw_params_set_periods_near( phandle, hw_params, &periods, &dir );
- if ( result < 0 ) {
- snd_pcm_close( phandle );
- errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting periods for device (" << name << "), " << snd_strerror( result ) << ".";
- errorText_ = errorStream_.str();
- return FAILURE;
- }
+ if ( options && options->flags & RTAUDIO_MINIMIZE_LATENCY ) periods = 2;
+ else periods = totalSize / *bufferSize;
+ // Even though the hardware might allow 1 buffer, it won't work reliably.
+ if ( periods < 2 ) periods = 2;
+ result = snd_pcm_hw_params_set_periods_near( phandle, hw_params, &periods, &dir );
+ if ( result < 0 ) {
+ snd_pcm_close( phandle );
+ errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting periods for device (" << name << "), " << snd_strerror( result ) << ".";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
- // If attempting to setup a duplex stream, the bufferSize parameter
- // MUST be the same in both directions!
- if ( stream_.mode == OUTPUT && mode == INPUT && *bufferSize != stream_.bufferSize ) {
- errorStream_ << "RtApiAlsa::probeDeviceOpen: system error setting buffer size for duplex stream on device (" << name << ").";
- errorText_ = errorStream_.str();
- return FAILURE;
- }
+ // If attempting to setup a duplex stream, the bufferSize parameter
+ // MUST be the same in both directions!
+ if ( stream_.mode == OUTPUT && mode == INPUT && *bufferSize != stream_.bufferSize ) {
+ errorStream_ << "RtApiAlsa::probeDeviceOpen: system error setting buffer size for duplex stream on device (" << name << ").";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
- stream_.bufferSize = *bufferSize;
+ stream_.bufferSize = *bufferSize;
- // Install the hardware configuration
- result = snd_pcm_hw_params( phandle, hw_params );
- if ( result < 0 ) {
- snd_pcm_close( phandle );
- errorStream_ << "RtApiAlsa::probeDeviceOpen: error installing hardware configuration on device (" << name << "), " << snd_strerror( result ) << ".";
- errorText_ = errorStream_.str();
- return FAILURE;
- }
+ // Install the hardware configuration
+ result = snd_pcm_hw_params( phandle, hw_params );
+ if ( result < 0 ) {
+ snd_pcm_close( phandle );
+ errorStream_ << "RtApiAlsa::probeDeviceOpen: error installing hardware configuration on device (" << name << "), " << snd_strerror( result ) << ".";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
#if defined(__RTAUDIO_DEBUG__)
- fprintf(stderr, "\nRtApiAlsa: dump hardware params after installation:\n\n");
- snd_pcm_hw_params_dump( hw_params, out );
+ fprintf(stderr, "\nRtApiAlsa: dump hardware params after installation:\n\n");
+ snd_pcm_hw_params_dump( hw_params, out );
#endif
- // Set the software configuration to fill buffers with zeros and prevent device stopping on xruns.
- snd_pcm_sw_params_t *sw_params = NULL;
- snd_pcm_sw_params_alloca( &sw_params );
- snd_pcm_sw_params_current( phandle, sw_params );
- snd_pcm_sw_params_set_start_threshold( phandle, sw_params, *bufferSize );
- snd_pcm_sw_params_set_stop_threshold( phandle, sw_params, ULONG_MAX );
- snd_pcm_sw_params_set_silence_threshold( phandle, sw_params, 0 );
-
- // The following two settings were suggested by Theo Veenker
- //snd_pcm_sw_params_set_avail_min( phandle, sw_params, *bufferSize );
- //snd_pcm_sw_params_set_xfer_align( phandle, sw_params, 1 );
-
- // here are two options for a fix
- //snd_pcm_sw_params_set_silence_size( phandle, sw_params, ULONG_MAX );
- snd_pcm_uframes_t val;
- snd_pcm_sw_params_get_boundary( sw_params, &val );
- snd_pcm_sw_params_set_silence_size( phandle, sw_params, val );
-
- result = snd_pcm_sw_params( phandle, sw_params );
- if ( result < 0 ) {
- snd_pcm_close( phandle );
- errorStream_ << "RtApiAlsa::probeDeviceOpen: error installing software configuration on device (" << name << "), " << snd_strerror( result ) << ".";
- errorText_ = errorStream_.str();
- return FAILURE;
- }
+ // Set the software configuration to fill buffers with zeros and prevent device stopping on xruns.
+ snd_pcm_sw_params_t *sw_params = NULL;
+ snd_pcm_sw_params_alloca( &sw_params );
+ snd_pcm_sw_params_current( phandle, sw_params );
+ snd_pcm_sw_params_set_start_threshold( phandle, sw_params, *bufferSize );
+ snd_pcm_sw_params_set_stop_threshold( phandle, sw_params, ULONG_MAX );
+ snd_pcm_sw_params_set_silence_threshold( phandle, sw_params, 0 );
+
+ // The following two settings were suggested by Theo Veenker
+ //snd_pcm_sw_params_set_avail_min( phandle, sw_params, *bufferSize );
+ //snd_pcm_sw_params_set_xfer_align( phandle, sw_params, 1 );
+
+ // here are two options for a fix
+ //snd_pcm_sw_params_set_silence_size( phandle, sw_params, ULONG_MAX );
+ snd_pcm_uframes_t val;
+ snd_pcm_sw_params_get_boundary( sw_params, &val );
+ snd_pcm_sw_params_set_silence_size( phandle, sw_params, val );
+
+ result = snd_pcm_sw_params( phandle, sw_params );
+ if ( result < 0 ) {
+ snd_pcm_close( phandle );
+ errorStream_ << "RtApiAlsa::probeDeviceOpen: error installing software configuration on device (" << name << "), " << snd_strerror( result ) << ".";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
#if defined(__RTAUDIO_DEBUG__)
- fprintf(stderr, "\nRtApiAlsa: dump software params after installation:\n\n");
- snd_pcm_sw_params_dump( sw_params, out );
+ fprintf(stderr, "\nRtApiAlsa: dump software params after installation:\n\n");
+ snd_pcm_sw_params_dump( sw_params, out );
#endif
- // Set flags for buffer conversion
- stream_.doConvertBuffer[mode] = false;
- if ( stream_.userFormat != stream_.deviceFormat[mode] )
- stream_.doConvertBuffer[mode] = true;
- if ( stream_.nUserChannels[mode] < stream_.nDeviceChannels[mode] )
- stream_.doConvertBuffer[mode] = true;
- if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] &&
- stream_.nUserChannels[mode] > 1 )
- stream_.doConvertBuffer[mode] = true;
-
- // Allocate the ApiHandle if necessary and then save.
- AlsaHandle *apiInfo = 0;
- if ( stream_.apiHandle == 0 ) {
- try {
- apiInfo = (AlsaHandle *) new AlsaHandle;
- }
- catch ( std::bad_alloc& ) {
- errorText_ = "RtApiAlsa::probeDeviceOpen: error allocating AlsaHandle memory.";
- goto error;
- }
-
- if ( pthread_cond_init( &apiInfo->runnable, NULL ) ) {
- errorText_ = "RtApiAlsa::probeDeviceOpen: error initializing pthread condition variable.";
- goto error;
- }
+ // Set flags for buffer conversion
+ stream_.doConvertBuffer[mode] = false;
+ if ( stream_.userFormat != stream_.deviceFormat[mode] )
+ stream_.doConvertBuffer[mode] = true;
+ if ( stream_.nUserChannels[mode] < stream_.nDeviceChannels[mode] )
+ stream_.doConvertBuffer[mode] = true;
+ if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] &&
+ stream_.nUserChannels[mode] > 1 )
+ stream_.doConvertBuffer[mode] = true;
- stream_.apiHandle = (void *) apiInfo;
- apiInfo->handles[0] = 0;
- apiInfo->handles[1] = 0;
+ // Allocate the ApiHandle if necessary and then save.
+ AlsaHandle *apiInfo = 0;
+ if ( stream_.apiHandle == 0 ) {
+ try {
+ apiInfo = (AlsaHandle *) new AlsaHandle;
}
- else {
- apiInfo = (AlsaHandle *) stream_.apiHandle;
+ catch ( std::bad_alloc& ) {
+ errorText_ = "RtApiAlsa::probeDeviceOpen: error allocating AlsaHandle memory.";
+ goto error;
}
- apiInfo->handles[mode] = phandle;
- // Allocate necessary internal buffers.
- unsigned long bufferBytes;
- bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat );
- stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 );
- if ( stream_.userBuffer[mode] == NULL ) {
- errorText_ = "RtApiAlsa::probeDeviceOpen: error allocating user buffer memory.";
+ if ( pthread_cond_init( &apiInfo->runnable, NULL ) ) {
+ errorText_ = "RtApiAlsa::probeDeviceOpen: error initializing pthread condition variable.";
goto error;
}
- if ( stream_.doConvertBuffer[mode] ) {
-
- bool makeBuffer = true;
- bufferBytes = stream_.nDeviceChannels[mode] * formatBytes( stream_.deviceFormat[mode] );
- if ( mode == INPUT ) {
- if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) {
- unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] );
- if ( bufferBytes <= bytesOut ) makeBuffer = false;
- }
- }
-
- if ( makeBuffer ) {
- bufferBytes *= *bufferSize;
- if ( stream_.deviceBuffer ) free( stream_.deviceBuffer );
- stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 );
- if ( stream_.deviceBuffer == NULL ) {
- errorText_ = "RtApiAlsa::probeDeviceOpen: error allocating device buffer memory.";
- goto error;
- }
- }
- }
+ stream_.apiHandle = (void *) apiInfo;
+ apiInfo->handles[0] = 0;
+ apiInfo->handles[1] = 0;
+ }
+ else {
+ apiInfo = (AlsaHandle *) stream_.apiHandle;
+ }
+ apiInfo->handles[mode] = phandle;
- stream_.sampleRate = sampleRate;
- stream_.nBuffers = periods;
- stream_.device[mode] = device;
- stream_.state = STREAM_STOPPED;
+ // Allocate necessary internal buffers.
+ unsigned long bufferBytes;
+ bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat );
+ stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 );
+ if ( stream_.userBuffer[mode] == NULL ) {
+ errorText_ = "RtApiAlsa::probeDeviceOpen: error allocating user buffer memory.";
+ goto error;
+ }
- // Setup the buffer conversion information structure.
- if ( stream_.doConvertBuffer[mode] ) setConvertInfo( mode, firstChannel );
+ if ( stream_.doConvertBuffer[mode] ) {
- // Setup thread if necessary.
- if ( stream_.mode == OUTPUT && mode == INPUT ) {
- // We had already set up an output stream.
- stream_.mode = DUPLEX;
- // Link the streams if possible.
- apiInfo->synchronized = false;
- if ( snd_pcm_link( apiInfo->handles[0], apiInfo->handles[1] ) == 0 )
- apiInfo->synchronized = true;
- else {
- errorText_ = "RtApiAlsa::probeDeviceOpen: unable to synchronize input and output devices.";
- error( RtError::WARNING );
+ bool makeBuffer = true;
+ bufferBytes = stream_.nDeviceChannels[mode] * formatBytes( stream_.deviceFormat[mode] );
+ if ( mode == INPUT ) {
+ if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) {
+ unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] );
+ if ( bufferBytes <= bytesOut ) makeBuffer = false;
}
}
- else {
- stream_.mode = mode;
-
- // Setup callback thread.
- stream_.callbackInfo.object = (void *) this;
-
- // Set the thread attributes for joinable and realtime scheduling
- // priority (optional). The higher priority will only take affect
- // if the program is run as root or suid. Note, under Linux
- // processes with CAP_SYS_NICE privilege, a user can change
- // scheduling policy and priority (thus need not be root). See
- // POSIX "capabilities".
- pthread_attr_t attr;
- pthread_attr_init( &attr );
- pthread_attr_setdetachstate( &attr, PTHREAD_CREATE_JOINABLE );
-#ifdef SCHED_RR // Undefined with some OSes (eg: NetBSD 1.6.x with GNU Pthread)
- if ( options && options->flags & RTAUDIO_SCHEDULE_REALTIME ) {
- struct sched_param param;
- int priority = options->priority;
- int min = sched_get_priority_min( SCHED_RR );
- int max = sched_get_priority_max( SCHED_RR );
- if ( priority < min ) priority = min;
- else if ( priority > max ) priority = max;
- param.sched_priority = priority;
- pthread_attr_setschedparam( &attr, ¶m );
- pthread_attr_setschedpolicy( &attr, SCHED_RR );
- }
- else
- pthread_attr_setschedpolicy( &attr, SCHED_OTHER );
-#else
- pthread_attr_setschedpolicy( &attr, SCHED_OTHER );
-#endif
- stream_.callbackInfo.isRunning = true;
- result = pthread_create( &stream_.callbackInfo.thread, &attr, alsaCallbackHandler, &stream_.callbackInfo );
- pthread_attr_destroy( &attr );
- if ( result ) {
- stream_.callbackInfo.isRunning = false;
- errorText_ = "RtApiAlsa::error creating callback thread!";
+ if ( makeBuffer ) {
+ bufferBytes *= *bufferSize;
+ if ( stream_.deviceBuffer ) free( stream_.deviceBuffer );
+ stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 );
+ if ( stream_.deviceBuffer == NULL ) {
+ errorText_ = "RtApiAlsa::probeDeviceOpen: error allocating device buffer memory.";
goto error;
}
}
-
- return SUCCESS;
-
- error:
- if ( apiInfo ) {
- pthread_cond_destroy( &apiInfo->runnable );
- if ( apiInfo->handles[0] ) snd_pcm_close( apiInfo->handles[0] );
- if ( apiInfo->handles[1] ) snd_pcm_close( apiInfo->handles[1] );
- delete apiInfo;
- stream_.apiHandle = 0;
- }
-
- for ( int i=0; i<2; i++ ) {
- if ( stream_.userBuffer[i] ) {
- free( stream_.userBuffer[i] );
- stream_.userBuffer[i] = 0;
- }
- }
-
- if ( stream_.deviceBuffer ) {
- free( stream_.deviceBuffer );
- stream_.deviceBuffer = 0;
- }
-
- return FAILURE;
}
- void RtApiAlsa :: closeStream()
- {
- if ( stream_.state == STREAM_CLOSED ) {
- errorText_ = "RtApiAlsa::closeStream(): no open stream to close!";
+ stream_.sampleRate = sampleRate;
+ stream_.nBuffers = periods;
+ stream_.device[mode] = device;
+ stream_.state = STREAM_STOPPED;
+
+ // Setup the buffer conversion information structure.
+ if ( stream_.doConvertBuffer[mode] ) setConvertInfo( mode, firstChannel );
+
+ // Setup thread if necessary.
+ if ( stream_.mode == OUTPUT && mode == INPUT ) {
+ // We had already set up an output stream.
+ stream_.mode = DUPLEX;
+ // Link the streams if possible.
+ apiInfo->synchronized = false;
+ if ( snd_pcm_link( apiInfo->handles[0], apiInfo->handles[1] ) == 0 )
+ apiInfo->synchronized = true;
+ else {
+ errorText_ = "RtApiAlsa::probeDeviceOpen: unable to synchronize input and output devices.";
error( RtError::WARNING );
- return;
}
+ }
+ else {
+ stream_.mode = mode;
- AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle;
- stream_.callbackInfo.isRunning = false;
- MUTEX_LOCK( &stream_.mutex );
- if ( stream_.state == STREAM_STOPPED )
- pthread_cond_signal( &apiInfo->runnable );
- MUTEX_UNLOCK( &stream_.mutex );
- pthread_join( stream_.callbackInfo.thread, NULL );
+ // Setup callback thread.
+ stream_.callbackInfo.object = (void *) this;
- if ( stream_.state == STREAM_RUNNING ) {
- stream_.state = STREAM_STOPPED;
- if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX )
- snd_pcm_drop( apiInfo->handles[0] );
- if ( stream_.mode == INPUT || stream_.mode == DUPLEX )
- snd_pcm_drop( apiInfo->handles[1] );
+ // Set the thread attributes for joinable and realtime scheduling
+ // priority (optional). The higher priority will only take affect
+ // if the program is run as root or suid. Note, under Linux
+ // processes with CAP_SYS_NICE privilege, a user can change
+ // scheduling policy and priority (thus need not be root). See
+ // POSIX "capabilities".
+ pthread_attr_t attr;
+ pthread_attr_init( &attr );
+ pthread_attr_setdetachstate( &attr, PTHREAD_CREATE_JOINABLE );
+#ifdef SCHED_RR // Undefined with some OSes (eg: NetBSD 1.6.x with GNU Pthread)
+ if ( options && options->flags & RTAUDIO_SCHEDULE_REALTIME ) {
+ struct sched_param param;
+ int priority = options->priority;
+ int min = sched_get_priority_min( SCHED_RR );
+ int max = sched_get_priority_max( SCHED_RR );
+ if ( priority < min ) priority = min;
+ else if ( priority > max ) priority = max;
+ param.sched_priority = priority;
+ pthread_attr_setschedparam( &attr, ¶m );
+ pthread_attr_setschedpolicy( &attr, SCHED_RR );
}
+ else
+ pthread_attr_setschedpolicy( &attr, SCHED_OTHER );
+#else
+ pthread_attr_setschedpolicy( &attr, SCHED_OTHER );
+#endif
- if ( apiInfo ) {
- pthread_cond_destroy( &apiInfo->runnable );
- if ( apiInfo->handles[0] ) snd_pcm_close( apiInfo->handles[0] );
- if ( apiInfo->handles[1] ) snd_pcm_close( apiInfo->handles[1] );
- delete apiInfo;
- stream_.apiHandle = 0;
+ stream_.callbackInfo.isRunning = true;
+ result = pthread_create( &stream_.callbackInfo.thread, &attr, alsaCallbackHandler, &stream_.callbackInfo );
+ pthread_attr_destroy( &attr );
+ if ( result ) {
+ stream_.callbackInfo.isRunning = false;
+ errorText_ = "RtApiAlsa::error creating callback thread!";
+ goto error;
}
+ }
- for ( int i=0; i<2; i++ ) {
- if ( stream_.userBuffer[i] ) {
- free( stream_.userBuffer[i] );
- stream_.userBuffer[i] = 0;
- }
- }
+ return SUCCESS;
+
+ error:
+ if ( apiInfo ) {
+ pthread_cond_destroy( &apiInfo->runnable );
+ if ( apiInfo->handles[0] ) snd_pcm_close( apiInfo->handles[0] );
+ if ( apiInfo->handles[1] ) snd_pcm_close( apiInfo->handles[1] );
+ delete apiInfo;
+ stream_.apiHandle = 0;
+ }
- if ( stream_.deviceBuffer ) {
- free( stream_.deviceBuffer );
- stream_.deviceBuffer = 0;
+ for ( int i=0; i<2; i++ ) {
+ if ( stream_.userBuffer[i] ) {
+ free( stream_.userBuffer[i] );
+ stream_.userBuffer[i] = 0;
}
+ }
- stream_.mode = UNINITIALIZED;
- stream_.state = STREAM_CLOSED;
+ if ( stream_.deviceBuffer ) {
+ free( stream_.deviceBuffer );
+ stream_.deviceBuffer = 0;
}
- void RtApiAlsa :: startStream()
- {
- // This method calls snd_pcm_prepare if the device isn't already in that state.
+ return FAILURE;
+}
- verifyStream();
- if ( stream_.state == STREAM_RUNNING ) {
- errorText_ = "RtApiAlsa::startStream(): the stream is already running!";
- error( RtError::WARNING );
- return;
- }
+void RtApiAlsa :: closeStream()
+{
+ if ( stream_.state == STREAM_CLOSED ) {
+ errorText_ = "RtApiAlsa::closeStream(): no open stream to close!";
+ error( RtError::WARNING );
+ return;
+ }
- MUTEX_LOCK( &stream_.mutex );
+ AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle;
+ stream_.callbackInfo.isRunning = false;
+ MUTEX_LOCK( &stream_.mutex );
+ if ( stream_.state == STREAM_STOPPED )
+ pthread_cond_signal( &apiInfo->runnable );
+ MUTEX_UNLOCK( &stream_.mutex );
+ pthread_join( stream_.callbackInfo.thread, NULL );
- int result = 0;
- snd_pcm_state_t state;
- AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle;
- snd_pcm_t **handle = (snd_pcm_t **) apiInfo->handles;
- if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
- state = snd_pcm_state( handle[0] );
- if ( state != SND_PCM_STATE_PREPARED ) {
- result = snd_pcm_prepare( handle[0] );
- if ( result < 0 ) {
- errorStream_ << "RtApiAlsa::startStream: error preparing output pcm device, " << snd_strerror( result ) << ".";
- errorText_ = errorStream_.str();
- goto unlock;
- }
- }
- }
+ if ( stream_.state == STREAM_RUNNING ) {
+ stream_.state = STREAM_STOPPED;
+ if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX )
+ snd_pcm_drop( apiInfo->handles[0] );
+ if ( stream_.mode == INPUT || stream_.mode == DUPLEX )
+ snd_pcm_drop( apiInfo->handles[1] );
+ }
- if ( ( stream_.mode == INPUT || stream_.mode == DUPLEX ) && !apiInfo->synchronized ) {
- state = snd_pcm_state( handle[1] );
- if ( state != SND_PCM_STATE_PREPARED ) {
- result = snd_pcm_prepare( handle[1] );
- if ( result < 0 ) {
- errorStream_ << "RtApiAlsa::startStream: error preparing input pcm device, " << snd_strerror( result ) << ".";
- errorText_ = errorStream_.str();
- goto unlock;
- }
- }
+ if ( apiInfo ) {
+ pthread_cond_destroy( &apiInfo->runnable );
+ if ( apiInfo->handles[0] ) snd_pcm_close( apiInfo->handles[0] );
+ if ( apiInfo->handles[1] ) snd_pcm_close( apiInfo->handles[1] );
+ delete apiInfo;
+ stream_.apiHandle = 0;
+ }
+
+ for ( int i=0; i<2; i++ ) {
+ if ( stream_.userBuffer[i] ) {
+ free( stream_.userBuffer[i] );
+ stream_.userBuffer[i] = 0;
}
+ }
- stream_.state = STREAM_RUNNING;
+ if ( stream_.deviceBuffer ) {
+ free( stream_.deviceBuffer );
+ stream_.deviceBuffer = 0;
+ }
- unlock:
- MUTEX_UNLOCK( &stream_.mutex );
+ stream_.mode = UNINITIALIZED;
+ stream_.state = STREAM_CLOSED;
+}
- pthread_cond_signal( &apiInfo->runnable );
+void RtApiAlsa :: startStream()
+{
+ // This method calls snd_pcm_prepare if the device isn't already in that state.
- if ( result >= 0 ) return;
- error( RtError::SYSTEM_ERROR );
+ verifyStream();
+ if ( stream_.state == STREAM_RUNNING ) {
+ errorText_ = "RtApiAlsa::startStream(): the stream is already running!";
+ error( RtError::WARNING );
+ return;
}
- void RtApiAlsa :: stopStream()
- {
- verifyStream();
- if ( stream_.state == STREAM_STOPPED ) {
- errorText_ = "RtApiAlsa::stopStream(): the stream is already stopped!";
- error( RtError::WARNING );
- return;
- }
-
- // Change the state before the lock to improve shutdown response.
- stream_.state = STREAM_STOPPED;
- MUTEX_LOCK( &stream_.mutex );
+ MUTEX_LOCK( &stream_.mutex );
- int result = 0;
- AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle;
- snd_pcm_t **handle = (snd_pcm_t **) apiInfo->handles;
- if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
- if ( apiInfo->synchronized )
- result = snd_pcm_drop( handle[0] );
- else
- result = snd_pcm_drain( handle[0] );
+ int result = 0;
+ snd_pcm_state_t state;
+ AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle;
+ snd_pcm_t **handle = (snd_pcm_t **) apiInfo->handles;
+ if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
+ state = snd_pcm_state( handle[0] );
+ if ( state != SND_PCM_STATE_PREPARED ) {
+ result = snd_pcm_prepare( handle[0] );
if ( result < 0 ) {
- errorStream_ << "RtApiAlsa::stopStream: error draining output pcm device, " << snd_strerror( result ) << ".";
+ errorStream_ << "RtApiAlsa::startStream: error preparing output pcm device, " << snd_strerror( result ) << ".";
errorText_ = errorStream_.str();
goto unlock;
}
}
+ }
- if ( ( stream_.mode == INPUT || stream_.mode == DUPLEX ) && !apiInfo->synchronized ) {
- result = snd_pcm_drop( handle[1] );
+ if ( ( stream_.mode == INPUT || stream_.mode == DUPLEX ) && !apiInfo->synchronized ) {
+ state = snd_pcm_state( handle[1] );
+ if ( state != SND_PCM_STATE_PREPARED ) {
+ result = snd_pcm_prepare( handle[1] );
if ( result < 0 ) {
- errorStream_ << "RtApiAlsa::stopStream: error stopping input pcm device, " << snd_strerror( result ) << ".";
+ errorStream_ << "RtApiAlsa::startStream: error preparing input pcm device, " << snd_strerror( result ) << ".";
errorText_ = errorStream_.str();
goto unlock;
}
}
+ }
- unlock:
- MUTEX_UNLOCK( &stream_.mutex );
+ stream_.state = STREAM_RUNNING;
- if ( result >= 0 ) return;
- error( RtError::SYSTEM_ERROR );
+ unlock:
+ MUTEX_UNLOCK( &stream_.mutex );
+
+ pthread_cond_signal( &apiInfo->runnable );
+
+ if ( result >= 0 ) return;
+ error( RtError::SYSTEM_ERROR );
+}
+
+void RtApiAlsa :: stopStream()
+{
+ verifyStream();
+ if ( stream_.state == STREAM_STOPPED ) {
+ errorText_ = "RtApiAlsa::stopStream(): the stream is already stopped!";
+ error( RtError::WARNING );
+ return;
}
- void RtApiAlsa :: abortStream()
- {
- verifyStream();
- if ( stream_.state == STREAM_STOPPED ) {
- errorText_ = "RtApiAlsa::abortStream(): the stream is already stopped!";
- error( RtError::WARNING );
- return;
- }
+ MUTEX_LOCK( &stream_.mutex );
- // Change the state before the lock to improve shutdown response.
- stream_.state = STREAM_STOPPED;
- MUTEX_LOCK( &stream_.mutex );
+ if ( stream_.state == STREAM_STOPPED ) {
+ MUTEX_UNLOCK( &stream_.mutex );
+ return;
+ }
- int result = 0;
- AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle;
- snd_pcm_t **handle = (snd_pcm_t **) apiInfo->handles;
- if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
+ int result = 0;
+ AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle;
+ snd_pcm_t **handle = (snd_pcm_t **) apiInfo->handles;
+ if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
+ if ( apiInfo->synchronized )
result = snd_pcm_drop( handle[0] );
- if ( result < 0 ) {
- errorStream_ << "RtApiAlsa::abortStream: error aborting output pcm device, " << snd_strerror( result ) << ".";
- errorText_ = errorStream_.str();
- goto unlock;
- }
+ else
+ result = snd_pcm_drain( handle[0] );
+ if ( result < 0 ) {
+ errorStream_ << "RtApiAlsa::stopStream: error draining output pcm device, " << snd_strerror( result ) << ".";
+ errorText_ = errorStream_.str();
+ goto unlock;
}
+ }
- if ( ( stream_.mode == INPUT || stream_.mode == DUPLEX ) && !apiInfo->synchronized ) {
- result = snd_pcm_drop( handle[1] );
- if ( result < 0 ) {
- errorStream_ << "RtApiAlsa::abortStream: error aborting input pcm device, " << snd_strerror( result ) << ".";
- errorText_ = errorStream_.str();
- goto unlock;
- }
+ if ( ( stream_.mode == INPUT || stream_.mode == DUPLEX ) && !apiInfo->synchronized ) {
+ result = snd_pcm_drop( handle[1] );
+ if ( result < 0 ) {
+ errorStream_ << "RtApiAlsa::stopStream: error stopping input pcm device, " << snd_strerror( result ) << ".";
+ errorText_ = errorStream_.str();
+ goto unlock;
}
+ }
- unlock:
- MUTEX_UNLOCK( &stream_.mutex );
+ unlock:
+ stream_.state = STREAM_STOPPED;
+ MUTEX_UNLOCK( &stream_.mutex );
+
+ if ( result >= 0 ) return;
+ error( RtError::SYSTEM_ERROR );
+}
- if ( result >= 0 ) return;
- error( RtError::SYSTEM_ERROR );
+void RtApiAlsa :: abortStream()
+{
+ verifyStream();
+ if ( stream_.state == STREAM_STOPPED ) {
+ errorText_ = "RtApiAlsa::abortStream(): the stream is already stopped!";
+ error( RtError::WARNING );
+ return;
}
- void RtApiAlsa :: callbackEvent()
- {
- AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle;
- if ( stream_.state == STREAM_STOPPED ) {
- MUTEX_LOCK( &stream_.mutex );
- pthread_cond_wait( &apiInfo->runnable, &stream_.mutex );
- if ( stream_.state != STREAM_RUNNING ) {
- MUTEX_UNLOCK( &stream_.mutex );
- return;
- }
- MUTEX_UNLOCK( &stream_.mutex );
- }
+ MUTEX_LOCK( &stream_.mutex );
- if ( stream_.state == STREAM_CLOSED ) {
- errorText_ = "RtApiAlsa::callbackEvent(): the stream is closed ... this shouldn't happen!";
- error( RtError::WARNING );
- return;
- }
+ if ( stream_.state == STREAM_STOPPED ) {
+ MUTEX_UNLOCK( &stream_.mutex );
+ return;
+ }
- int doStopStream = 0;
- RtAudioCallback callback = (RtAudioCallback) stream_.callbackInfo.callback;
- double streamTime = getStreamTime();
- RtAudioStreamStatus status = 0;
- if ( stream_.mode != INPUT && apiInfo->xrun[0] == true ) {
- status |= RTAUDIO_OUTPUT_UNDERFLOW;
- apiInfo->xrun[0] = false;
+ int result = 0;
+ AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle;
+ snd_pcm_t **handle = (snd_pcm_t **) apiInfo->handles;
+ if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
+ result = snd_pcm_drop( handle[0] );
+ if ( result < 0 ) {
+ errorStream_ << "RtApiAlsa::abortStream: error aborting output pcm device, " << snd_strerror( result ) << ".";
+ errorText_ = errorStream_.str();
+ goto unlock;
}
- if ( stream_.mode != OUTPUT && apiInfo->xrun[1] == true ) {
- status |= RTAUDIO_INPUT_OVERFLOW;
- apiInfo->xrun[1] = false;
+ }
+
+ if ( ( stream_.mode == INPUT || stream_.mode == DUPLEX ) && !apiInfo->synchronized ) {
+ result = snd_pcm_drop( handle[1] );
+ if ( result < 0 ) {
+ errorStream_ << "RtApiAlsa::abortStream: error aborting input pcm device, " << snd_strerror( result ) << ".";
+ errorText_ = errorStream_.str();
+ goto unlock;
}
- doStopStream = callback( stream_.userBuffer[0], stream_.userBuffer[1],
- stream_.bufferSize, streamTime, status, stream_.callbackInfo.userData );
+ }
- if ( doStopStream == 2 ) {
- abortStream();
+ unlock:
+ stream_.state = STREAM_STOPPED;
+ MUTEX_UNLOCK( &stream_.mutex );
+
+ if ( result >= 0 ) return;
+ error( RtError::SYSTEM_ERROR );
+}
+
+void RtApiAlsa :: callbackEvent()
+{
+ AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle;
+ if ( stream_.state == STREAM_STOPPED ) {
+ MUTEX_LOCK( &stream_.mutex );
+ pthread_cond_wait( &apiInfo->runnable, &stream_.mutex );
+ if ( stream_.state != STREAM_RUNNING ) {
+ MUTEX_UNLOCK( &stream_.mutex );
return;
}
+ MUTEX_UNLOCK( &stream_.mutex );
+ }
- MUTEX_LOCK( &stream_.mutex );
+ if ( stream_.state == STREAM_CLOSED ) {
+ errorText_ = "RtApiAlsa::callbackEvent(): the stream is closed ... this shouldn't happen!";
+ error( RtError::WARNING );
+ return;
+ }
- // The state might change while waiting on a mutex.
- if ( stream_.state == STREAM_STOPPED ) goto unlock;
+ int doStopStream = 0;
+ RtAudioCallback callback = (RtAudioCallback) stream_.callbackInfo.callback;
+ double streamTime = getStreamTime();
+ RtAudioStreamStatus status = 0;
+ if ( stream_.mode != INPUT && apiInfo->xrun[0] == true ) {
+ status |= RTAUDIO_OUTPUT_UNDERFLOW;
+ apiInfo->xrun[0] = false;
+ }
+ if ( stream_.mode != OUTPUT && apiInfo->xrun[1] == true ) {
+ status |= RTAUDIO_INPUT_OVERFLOW;
+ apiInfo->xrun[1] = false;
+ }
+ doStopStream = callback( stream_.userBuffer[0], stream_.userBuffer[1],
+ stream_.bufferSize, streamTime, status, stream_.callbackInfo.userData );
- int result;
- char *buffer;
- int channels;
- snd_pcm_t **handle;
- snd_pcm_sframes_t frames;
- RtAudioFormat format;
- handle = (snd_pcm_t **) apiInfo->handles;
+ if ( doStopStream == 2 ) {
+ abortStream();
+ return;
+ }
- if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {
+ MUTEX_LOCK( &stream_.mutex );
- // Setup parameters.
- if ( stream_.doConvertBuffer[1] ) {
- buffer = stream_.deviceBuffer;
- channels = stream_.nDeviceChannels[1];
- format = stream_.deviceFormat[1];
- }
- else {
- buffer = stream_.userBuffer[1];
- channels = stream_.nUserChannels[1];
- format = stream_.userFormat;
- }
+ // The state might change while waiting on a mutex.
+ if ( stream_.state == STREAM_STOPPED ) goto unlock;
- // Read samples from device in interleaved/non-interleaved format.
- if ( stream_.deviceInterleaved[1] )
- result = snd_pcm_readi( handle[1], buffer, stream_.bufferSize );
- else {
- void *bufs[channels];
- size_t offset = stream_.bufferSize * formatBytes( format );
- for ( int i=0; i<channels; i++ )
- bufs[i] = (void *) (buffer + (i * offset));
- result = snd_pcm_readn( handle[1], bufs, stream_.bufferSize );
- }
+ int result;
+ char *buffer;
+ int channels;
+ snd_pcm_t **handle;
+ snd_pcm_sframes_t frames;
+ RtAudioFormat format;
+ handle = (snd_pcm_t **) apiInfo->handles;
- if ( result < (int) stream_.bufferSize ) {
- // Either an error or overrun occured.
- if ( result == -EPIPE ) {
- snd_pcm_state_t state = snd_pcm_state( handle[1] );
- if ( state == SND_PCM_STATE_XRUN ) {
- apiInfo->xrun[1] = true;
- result = snd_pcm_prepare( handle[1] );
- if ( result < 0 ) {
- errorStream_ << "RtApiAlsa::callbackEvent: error preparing device after overrun, " << snd_strerror( result ) << ".";
- errorText_ = errorStream_.str();
- }
- }
- else {
- errorStream_ << "RtApiAlsa::callbackEvent: error, current state is " << snd_pcm_state_name( state ) << ", " << snd_strerror( result ) << ".";
+ if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {
+
+ // Setup parameters.
+ if ( stream_.doConvertBuffer[1] ) {
+ buffer = stream_.deviceBuffer;
+ channels = stream_.nDeviceChannels[1];
+ format = stream_.deviceFormat[1];
+ }
+ else {
+ buffer = stream_.userBuffer[1];
+ channels = stream_.nUserChannels[1];
+ format = stream_.userFormat;
+ }
+
+ // Read samples from device in interleaved/non-interleaved format.
+ if ( stream_.deviceInterleaved[1] )
+ result = snd_pcm_readi( handle[1], buffer, stream_.bufferSize );
+ else {
+ void *bufs[channels];
+ size_t offset = stream_.bufferSize * formatBytes( format );
+ for ( int i=0; i<channels; i++ )
+ bufs[i] = (void *) (buffer + (i * offset));
+ result = snd_pcm_readn( handle[1], bufs, stream_.bufferSize );
+ }
+
+ if ( result < (int) stream_.bufferSize ) {
+ // Either an error or overrun occured.
+ if ( result == -EPIPE ) {
+ snd_pcm_state_t state = snd_pcm_state( handle[1] );
+ if ( state == SND_PCM_STATE_XRUN ) {
+ apiInfo->xrun[1] = true;
+ result = snd_pcm_prepare( handle[1] );
+ if ( result < 0 ) {
+ errorStream_ << "RtApiAlsa::callbackEvent: error preparing device after overrun, " << snd_strerror( result ) << ".";
errorText_ = errorStream_.str();
}
}
else {
- errorStream_ << "RtApiAlsa::callbackEvent: audio read error, " << snd_strerror( result ) << ".";
+ errorStream_ << "RtApiAlsa::callbackEvent: error, current state is " << snd_pcm_state_name( state ) << ", " << snd_strerror( result ) << ".";
errorText_ = errorStream_.str();
}
- error( RtError::WARNING );
- goto tryOutput;
}
+ else {
+ errorStream_ << "RtApiAlsa::callbackEvent: audio read error, " << snd_strerror( result ) << ".";
+ errorText_ = errorStream_.str();
+ }
+ error( RtError::WARNING );
+ goto tryOutput;
+ }
- // Do byte swapping if necessary.
- if ( stream_.doByteSwap[1] )
- byteSwapBuffer( buffer, stream_.bufferSize * channels, format );
-
- // Do buffer conversion if necessary.
- if ( stream_.doConvertBuffer[1] )
- convertBuffer( stream_.userBuffer[1], stream_.deviceBuffer, stream_.convertInfo[1] );
+ // Do byte swapping if necessary.
+ if ( stream_.doByteSwap[1] )
+ byteSwapBuffer( buffer, stream_.bufferSize * channels, format );
- // Check stream latency
- result = snd_pcm_delay( handle[1], &frames );
- if ( result == 0 && frames > 0 ) stream_.latency[1] = frames;
- }
+ // Do buffer conversion if necessary.
+ if ( stream_.doConvertBuffer[1] )
+ convertBuffer( stream_.userBuffer[1], stream_.deviceBuffer, stream_.convertInfo[1] );
- tryOutput:
+ // Check stream latency
+ result = snd_pcm_delay( handle[1], &frames );
+ if ( result == 0 && frames > 0 ) stream_.latency[1] = frames;
+ }
- if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
+ tryOutput:
- // Setup parameters and do buffer conversion if necessary.
- if ( stream_.doConvertBuffer[0] ) {
- buffer = stream_.deviceBuffer;
- convertBuffer( buffer, stream_.userBuffer[0], stream_.convertInfo[0] );
- channels = stream_.nDeviceChannels[0];
- format = stream_.deviceFormat[0];
- }
- else {
- buffer = stream_.userBuffer[0];
- channels = stream_.nUserChannels[0];
- format = stream_.userFormat;
- }
+ if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
- // Do byte swapping if necessary.
- if ( stream_.doByteSwap[0] )
- byteSwapBuffer(buffer, stream_.bufferSize * channels, format);
+ // Setup parameters and do buffer conversion if necessary.
+ if ( stream_.doConvertBuffer[0] ) {
+ buffer = stream_.deviceBuffer;
+ convertBuffer( buffer, stream_.userBuffer[0], stream_.convertInfo[0] );
+ channels = stream_.nDeviceChannels[0];
+ format = stream_.deviceFormat[0];
+ }
+ else {
+ buffer = stream_.userBuffer[0];
+ channels = stream_.nUserChannels[0];
+ format = stream_.userFormat;
+ }
- // Write samples to device in interleaved/non-interleaved format.
- if ( stream_.deviceInterleaved[0] )
- result = snd_pcm_writei( handle[0], buffer, stream_.bufferSize );
- else {
- void *bufs[channels];
- size_t offset = stream_.bufferSize * formatBytes( format );
- for ( int i=0; i<channels; i++ )
- bufs[i] = (void *) (buffer + (i * offset));
- result = snd_pcm_writen( handle[0], bufs, stream_.bufferSize );
- }
+ // Do byte swapping if necessary.
+ if ( stream_.doByteSwap[0] )
+ byteSwapBuffer(buffer, stream_.bufferSize * channels, format);
- if ( result < (int) stream_.bufferSize ) {
- // Either an error or underrun occured.
- if ( result == -EPIPE ) {
- snd_pcm_state_t state = snd_pcm_state( handle[0] );
- if ( state == SND_PCM_STATE_XRUN ) {
- apiInfo->xrun[0] = true;
- result = snd_pcm_prepare( handle[0] );
- if ( result < 0 ) {
- errorStream_ << "RtApiAlsa::callbackEvent: error preparing device after underrun, " << snd_strerror( result ) << ".";
- errorText_ = errorStream_.str();
- }
- }
- else {
- errorStream_ << "RtApiAlsa::callbackEvent: error, current state is " << snd_pcm_state_name( state ) << ", " << snd_strerror( result ) << ".";
+ // Write samples to device in interleaved/non-interleaved format.
+ if ( stream_.deviceInterleaved[0] )
+ result = snd_pcm_writei( handle[0], buffer, stream_.bufferSize );
+ else {
+ void *bufs[channels];
+ size_t offset = stream_.bufferSize * formatBytes( format );
+ for ( int i=0; i<channels; i++ )
+ bufs[i] = (void *) (buffer + (i * offset));
+ result = snd_pcm_writen( handle[0], bufs, stream_.bufferSize );
+ }
+
+ if ( result < (int) stream_.bufferSize ) {
+ // Either an error or underrun occured.
+ if ( result == -EPIPE ) {
+ snd_pcm_state_t state = snd_pcm_state( handle[0] );
+ if ( state == SND_PCM_STATE_XRUN ) {
+ apiInfo->xrun[0] = true;
+ result = snd_pcm_prepare( handle[0] );
+ if ( result < 0 ) {
+ errorStream_ << "RtApiAlsa::callbackEvent: error preparing device after underrun, " << snd_strerror( result ) << ".";
errorText_ = errorStream_.str();
}
}
else {
- errorStream_ << "RtApiAlsa::callbackEvent: audio write error, " << snd_strerror( result ) << ".";
+ errorStream_ << "RtApiAlsa::callbackEvent: error, current state is " << snd_pcm_state_name( state ) << ", " << snd_strerror( result ) << ".";
errorText_ = errorStream_.str();
}
- error( RtError::WARNING );
- goto unlock;
}
-
- // Check stream latency
- result = snd_pcm_delay( handle[0], &frames );
- if ( result == 0 && frames > 0 ) stream_.latency[0] = frames;
+ else {
+ errorStream_ << "RtApiAlsa::callbackEvent: audio write error, " << snd_strerror( result ) << ".";
+ errorText_ = errorStream_.str();
+ }
+ error( RtError::WARNING );
+ goto unlock;
}
- unlock:
- MUTEX_UNLOCK( &stream_.mutex );
-
- RtApi::tickStreamTime();
- if ( doStopStream == 1 ) this->stopStream();
+ // Check stream latency
+ result = snd_pcm_delay( handle[0], &frames );
+ if ( result == 0 && frames > 0 ) stream_.latency[0] = frames;
}
- extern "C" void *alsaCallbackHandler( void *ptr )
- {
- CallbackInfo *info = (CallbackInfo *) ptr;
- RtApiAlsa *object = (RtApiAlsa *) info->object;
- bool *isRunning = &info->isRunning;
+ unlock:
+ MUTEX_UNLOCK( &stream_.mutex );
+
+ RtApi::tickStreamTime();
+ if ( doStopStream == 1 ) this->stopStream();
+}
- while ( *isRunning == true ) {
- pthread_testcancel();
- object->callbackEvent();
- }
+extern "C" void *alsaCallbackHandler( void *ptr )
+{
+ CallbackInfo *info = (CallbackInfo *) ptr;
+ RtApiAlsa *object = (RtApiAlsa *) info->object;
+ bool *isRunning = &info->isRunning;
- pthread_exit( NULL );
+ while ( *isRunning == true ) {
+ pthread_testcancel();
+ object->callbackEvent();
}
- //******************** End of __LINUX_ALSA__ *********************//
+ pthread_exit( NULL );
+}
+
+//******************** End of __LINUX_ALSA__ *********************//
#endif
#include <errno.h>
#include <math.h>
- extern "C" void *ossCallbackHandler(void * ptr);
+extern "C" void *ossCallbackHandler(void * ptr);
- // A structure to hold various information related to the OSS API
- // implementation.
- struct OssHandle {
- int id[2]; // device ids
- bool xrun[2];
- bool triggered;
- pthread_cond_t runnable;
-
- OssHandle()
- :triggered(false) { id[0] = 0; id[1] = 0; xrun[0] = false; xrun[1] = false; }
- };
+// A structure to hold various information related to the OSS API
+// implementation.
+struct OssHandle {
+ int id[2]; // device ids
+ bool xrun[2];
+ bool triggered;
+ pthread_cond_t runnable;
- RtApiOss :: RtApiOss()
- {
- // Nothing to do here.
- }
+ OssHandle()
+ :triggered(false) { id[0] = 0; id[1] = 0; xrun[0] = false; xrun[1] = false; }
+};
- RtApiOss :: ~RtApiOss()
- {
- if ( stream_.state != STREAM_CLOSED ) closeStream();
- }
+RtApiOss :: RtApiOss()
+{
+ // Nothing to do here.
+}
- unsigned int RtApiOss :: getDeviceCount( void )
- {
- int mixerfd = open( "/dev/mixer", O_RDWR, 0 );
- if ( mixerfd == -1 ) {
- errorText_ = "RtApiOss::getDeviceCount: error opening '/dev/mixer'.";
- error( RtError::WARNING );
- return 0;
- }
+RtApiOss :: ~RtApiOss()
+{
+ if ( stream_.state != STREAM_CLOSED ) closeStream();
+}
- oss_sysinfo sysinfo;
- if ( ioctl( mixerfd, SNDCTL_SYSINFO, &sysinfo ) == -1 ) {
- close( mixerfd );
- errorText_ = "RtApiOss::getDeviceCount: error getting sysinfo, OSS version >= 4.0 is required.";
- error( RtError::WARNING );
- return 0;
- }
+unsigned int RtApiOss :: getDeviceCount( void )
+{
+ int mixerfd = open( "/dev/mixer", O_RDWR, 0 );
+ if ( mixerfd == -1 ) {
+ errorText_ = "RtApiOss::getDeviceCount: error opening '/dev/mixer'.";
+ error( RtError::WARNING );
+ return 0;
+ }
+ oss_sysinfo sysinfo;
+ if ( ioctl( mixerfd, SNDCTL_SYSINFO, &sysinfo ) == -1 ) {
close( mixerfd );
- return sysinfo.numaudios;
+ errorText_ = "RtApiOss::getDeviceCount: error getting sysinfo, OSS version >= 4.0 is required.";
+ error( RtError::WARNING );
+ return 0;
}
- RtAudio::DeviceInfo RtApiOss :: getDeviceInfo( unsigned int device )
- {
- RtAudio::DeviceInfo info;
- info.probed = false;
+ close( mixerfd );
+ return sysinfo.numaudios;
+}
- int mixerfd = open( "/dev/mixer", O_RDWR, 0 );
- if ( mixerfd == -1 ) {
- errorText_ = "RtApiOss::getDeviceInfo: error opening '/dev/mixer'.";
- error( RtError::WARNING );
- return info;
- }
+RtAudio::DeviceInfo RtApiOss :: getDeviceInfo( unsigned int device )
+{
+ RtAudio::DeviceInfo info;
+ info.probed = false;
- oss_sysinfo sysinfo;
- int result = ioctl( mixerfd, SNDCTL_SYSINFO, &sysinfo );
- if ( result == -1 ) {
- close( mixerfd );
- errorText_ = "RtApiOss::getDeviceInfo: error getting sysinfo, OSS version >= 4.0 is required.";
- error( RtError::WARNING );
- return info;
- }
+ int mixerfd = open( "/dev/mixer", O_RDWR, 0 );
+ if ( mixerfd == -1 ) {
+ errorText_ = "RtApiOss::getDeviceInfo: error opening '/dev/mixer'.";
+ error( RtError::WARNING );
+ return info;
+ }
- unsigned nDevices = sysinfo.numaudios;
- if ( nDevices == 0 ) {
- close( mixerfd );
- errorText_ = "RtApiOss::getDeviceInfo: no devices found!";
- error( RtError::INVALID_USE );
- }
+ oss_sysinfo sysinfo;
+ int result = ioctl( mixerfd, SNDCTL_SYSINFO, &sysinfo );
+ if ( result == -1 ) {
+ close( mixerfd );
+ errorText_ = "RtApiOss::getDeviceInfo: error getting sysinfo, OSS version >= 4.0 is required.";
+ error( RtError::WARNING );
+ return info;
+ }
- if ( device >= nDevices ) {
- close( mixerfd );
- errorText_ = "RtApiOss::getDeviceInfo: device ID is invalid!";
- error( RtError::INVALID_USE );
- }
+ unsigned nDevices = sysinfo.numaudios;
+ if ( nDevices == 0 ) {
+ close( mixerfd );
+ errorText_ = "RtApiOss::getDeviceInfo: no devices found!";
+ error( RtError::INVALID_USE );
+ }
- oss_audioinfo ainfo;
- ainfo.dev = device;
- result = ioctl( mixerfd, SNDCTL_AUDIOINFO, &ainfo );
+ if ( device >= nDevices ) {
close( mixerfd );
- if ( result == -1 ) {
- errorStream_ << "RtApiOss::getDeviceInfo: error getting device (" << ainfo.name << ") info.";
- errorText_ = errorStream_.str();
- error( RtError::WARNING );
- return info;
- }
+ errorText_ = "RtApiOss::getDeviceInfo: device ID is invalid!";
+ error( RtError::INVALID_USE );
+ }
- // Probe channels
- if ( ainfo.caps & PCM_CAP_OUTPUT ) info.outputChannels = ainfo.max_channels;
- if ( ainfo.caps & PCM_CAP_INPUT ) info.inputChannels = ainfo.max_channels;
- if ( ainfo.caps & PCM_CAP_DUPLEX ) {
- if ( info.outputChannels > 0 && info.inputChannels > 0 && ainfo.caps & PCM_CAP_DUPLEX )
- info.duplexChannels = (info.outputChannels > info.inputChannels) ? info.inputChannels : info.outputChannels;
- }
-
- // Probe data formats ... do for input
- unsigned long mask = ainfo.iformats;
- if ( mask & AFMT_S16_LE || mask & AFMT_S16_BE )
- info.nativeFormats |= RTAUDIO_SINT16;
- if ( mask & AFMT_S8 )
- info.nativeFormats |= RTAUDIO_SINT8;
- if ( mask & AFMT_S32_LE || mask & AFMT_S32_BE )
- info.nativeFormats |= RTAUDIO_SINT32;
- if ( mask & AFMT_FLOAT )
- info.nativeFormats |= RTAUDIO_FLOAT32;
- if ( mask & AFMT_S24_LE || mask & AFMT_S24_BE )
- info.nativeFormats |= RTAUDIO_SINT24;
-
- // Check that we have at least one supported format
- if ( info.nativeFormats == 0 ) {
- errorStream_ << "RtApiOss::getDeviceInfo: device (" << ainfo.name << ") data format not supported by RtAudio.";
- errorText_ = errorStream_.str();
- error( RtError::WARNING );
- return info;
- }
+ oss_audioinfo ainfo;
+ ainfo.dev = device;
+ result = ioctl( mixerfd, SNDCTL_AUDIOINFO, &ainfo );
+ close( mixerfd );
+ if ( result == -1 ) {
+ errorStream_ << "RtApiOss::getDeviceInfo: error getting device (" << ainfo.name << ") info.";
+ errorText_ = errorStream_.str();
+ error( RtError::WARNING );
+ return info;
+ }
- // Probe the supported sample rates.
- info.sampleRates.clear();
- if ( ainfo.nrates ) {
- for ( unsigned int i=0; i<ainfo.nrates; i++ ) {
- for ( unsigned int k=0; k<MAX_SAMPLE_RATES; k++ ) {
- if ( ainfo.rates[i] == SAMPLE_RATES[k] ) {
- info.sampleRates.push_back( SAMPLE_RATES[k] );
- break;
- }
- }
- }
- }
- else {
- // Check min and max rate values;
+ // Probe channels
+ if ( ainfo.caps & PCM_CAP_OUTPUT ) info.outputChannels = ainfo.max_channels;
+ if ( ainfo.caps & PCM_CAP_INPUT ) info.inputChannels = ainfo.max_channels;
+ if ( ainfo.caps & PCM_CAP_DUPLEX ) {
+ if ( info.outputChannels > 0 && info.inputChannels > 0 && ainfo.caps & PCM_CAP_DUPLEX )
+ info.duplexChannels = (info.outputChannels > info.inputChannels) ? info.inputChannels : info.outputChannels;
+ }
+
+ // Probe data formats ... do for input
+ unsigned long mask = ainfo.iformats;
+ if ( mask & AFMT_S16_LE || mask & AFMT_S16_BE )
+ info.nativeFormats |= RTAUDIO_SINT16;
+ if ( mask & AFMT_S8 )
+ info.nativeFormats |= RTAUDIO_SINT8;
+ if ( mask & AFMT_S32_LE || mask & AFMT_S32_BE )
+ info.nativeFormats |= RTAUDIO_SINT32;
+ if ( mask & AFMT_FLOAT )
+ info.nativeFormats |= RTAUDIO_FLOAT32;
+ if ( mask & AFMT_S24_LE || mask & AFMT_S24_BE )
+ info.nativeFormats |= RTAUDIO_SINT24;
+
+ // Check that we have at least one supported format
+ if ( info.nativeFormats == 0 ) {
+ errorStream_ << "RtApiOss::getDeviceInfo: device (" << ainfo.name << ") data format not supported by RtAudio.";
+ errorText_ = errorStream_.str();
+ error( RtError::WARNING );
+ return info;
+ }
+
+ // Probe the supported sample rates.
+ info.sampleRates.clear();
+ if ( ainfo.nrates ) {
+ for ( unsigned int i=0; i<ainfo.nrates; i++ ) {
for ( unsigned int k=0; k<MAX_SAMPLE_RATES; k++ ) {
- if ( ainfo.min_rate <= (int) SAMPLE_RATES[k] && ainfo.max_rate >= (int) SAMPLE_RATES[k] )
+ if ( ainfo.rates[i] == SAMPLE_RATES[k] ) {
info.sampleRates.push_back( SAMPLE_RATES[k] );
+ break;
+ }
}
}
-
- if ( info.sampleRates.size() == 0 ) {
- errorStream_ << "RtApiOss::getDeviceInfo: no supported sample rates found for device (" << ainfo.name << ").";
- errorText_ = errorStream_.str();
- error( RtError::WARNING );
- }
- else {
- info.probed = true;
- info.name = ainfo.name;
+ }
+ else {
+ // Check min and max rate values;
+ for ( unsigned int k=0; k<MAX_SAMPLE_RATES; k++ ) {
+ if ( ainfo.min_rate <= (int) SAMPLE_RATES[k] && ainfo.max_rate >= (int) SAMPLE_RATES[k] )
+ info.sampleRates.push_back( SAMPLE_RATES[k] );
}
+ }
- return info;
+ if ( info.sampleRates.size() == 0 ) {
+ errorStream_ << "RtApiOss::getDeviceInfo: no supported sample rates found for device (" << ainfo.name << ").";
+ errorText_ = errorStream_.str();
+ error( RtError::WARNING );
+ }
+ else {
+ info.probed = true;
+ info.name = ainfo.name;
}
+ return info;
+}
- bool RtApiOss :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,
- unsigned int firstChannel, unsigned int sampleRate,
- RtAudioFormat format, unsigned int *bufferSize,
- RtAudio::StreamOptions *options )
- {
- int mixerfd = open( "/dev/mixer", O_RDWR, 0 );
- if ( mixerfd == -1 ) {
- errorText_ = "RtApiOss::probeDeviceOpen: error opening '/dev/mixer'.";
- return FAILURE;
- }
- oss_sysinfo sysinfo;
- int result = ioctl( mixerfd, SNDCTL_SYSINFO, &sysinfo );
- if ( result == -1 ) {
- close( mixerfd );
- errorText_ = "RtApiOss::probeDeviceOpen: error getting sysinfo, OSS version >= 4.0 is required.";
- return FAILURE;
- }
+bool RtApiOss :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,
+ unsigned int firstChannel, unsigned int sampleRate,
+ RtAudioFormat format, unsigned int *bufferSize,
+ RtAudio::StreamOptions *options )
+{
+ int mixerfd = open( "/dev/mixer", O_RDWR, 0 );
+ if ( mixerfd == -1 ) {
+ errorText_ = "RtApiOss::probeDeviceOpen: error opening '/dev/mixer'.";
+ return FAILURE;
+ }
- unsigned nDevices = sysinfo.numaudios;
- if ( nDevices == 0 ) {
- // This should not happen because a check is made before this function is called.
- close( mixerfd );
- errorText_ = "RtApiOss::probeDeviceOpen: no devices found!";
- return FAILURE;
- }
+ oss_sysinfo sysinfo;
+ int result = ioctl( mixerfd, SNDCTL_SYSINFO, &sysinfo );
+ if ( result == -1 ) {
+ close( mixerfd );
+ errorText_ = "RtApiOss::probeDeviceOpen: error getting sysinfo, OSS version >= 4.0 is required.";
+ return FAILURE;
+ }
- if ( device >= nDevices ) {
- // This should not happen because a check is made before this function is called.
- close( mixerfd );
- errorText_ = "RtApiOss::probeDeviceOpen: device ID is invalid!";
- return FAILURE;
- }
+ unsigned nDevices = sysinfo.numaudios;
+ if ( nDevices == 0 ) {
+ // This should not happen because a check is made before this function is called.
+ close( mixerfd );
+ errorText_ = "RtApiOss::probeDeviceOpen: no devices found!";
+ return FAILURE;
+ }
- oss_audioinfo ainfo;
- ainfo.dev = device;
- result = ioctl( mixerfd, SNDCTL_AUDIOINFO, &ainfo );
+ if ( device >= nDevices ) {
+ // This should not happen because a check is made before this function is called.
close( mixerfd );
- if ( result == -1 ) {
- errorStream_ << "RtApiOss::getDeviceInfo: error getting device (" << ainfo.name << ") info.";
- errorText_ = errorStream_.str();
- return FAILURE;
- }
+ errorText_ = "RtApiOss::probeDeviceOpen: device ID is invalid!";
+ return FAILURE;
+ }
- // Check if device supports input or output
- if ( ( mode == OUTPUT && !( ainfo.caps & PCM_CAP_OUTPUT ) ) ||
- ( mode == INPUT && !( ainfo.caps & PCM_CAP_INPUT ) ) ) {
- if ( mode == OUTPUT )
- errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") does not support output.";
- else
- errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") does not support input.";
- errorText_ = errorStream_.str();
- return FAILURE;
- }
+ oss_audioinfo ainfo;
+ ainfo.dev = device;
+ result = ioctl( mixerfd, SNDCTL_AUDIOINFO, &ainfo );
+ close( mixerfd );
+ if ( result == -1 ) {
+ errorStream_ << "RtApiOss::getDeviceInfo: error getting device (" << ainfo.name << ") info.";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
- int flags = 0;
- OssHandle *handle = (OssHandle *) stream_.apiHandle;
+ // Check if device supports input or output
+ if ( ( mode == OUTPUT && !( ainfo.caps & PCM_CAP_OUTPUT ) ) ||
+ ( mode == INPUT && !( ainfo.caps & PCM_CAP_INPUT ) ) ) {
if ( mode == OUTPUT )
- flags |= O_WRONLY;
- else { // mode == INPUT
- if (stream_.mode == OUTPUT && stream_.device[0] == device) {
- // We just set the same device for playback ... close and reopen for duplex (OSS only).
- close( handle->id[0] );
- handle->id[0] = 0;
- if ( !( ainfo.caps & PCM_CAP_DUPLEX ) ) {
- errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") does not support duplex mode.";
- errorText_ = errorStream_.str();
- return FAILURE;
- }
- // Check that the number previously set channels is the same.
- if ( stream_.nUserChannels[0] != channels ) {
- errorStream_ << "RtApiOss::probeDeviceOpen: input/output channels must be equal for OSS duplex device (" << ainfo.name << ").";
- errorText_ = errorStream_.str();
- return FAILURE;
- }
- flags |= O_RDWR;
+ errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") does not support output.";
+ else
+ errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") does not support input.";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
+
+ int flags = 0;
+ OssHandle *handle = (OssHandle *) stream_.apiHandle;
+ if ( mode == OUTPUT )
+ flags |= O_WRONLY;
+ else { // mode == INPUT
+ if (stream_.mode == OUTPUT && stream_.device[0] == device) {
+ // We just set the same device for playback ... close and reopen for duplex (OSS only).
+ close( handle->id[0] );
+ handle->id[0] = 0;
+ if ( !( ainfo.caps & PCM_CAP_DUPLEX ) ) {
+ errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") does not support duplex mode.";
+ errorText_ = errorStream_.str();
+ return FAILURE;
}
- else
- flags |= O_RDONLY;
+ // Check that the number previously set channels is the same.
+ if ( stream_.nUserChannels[0] != channels ) {
+ errorStream_ << "RtApiOss::probeDeviceOpen: input/output channels must be equal for OSS duplex device (" << ainfo.name << ").";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
+ flags |= O_RDWR;
}
+ else
+ flags |= O_RDONLY;
+ }
- // Set exclusive access if specified.
- if ( options && options->flags & RTAUDIO_HOG_DEVICE ) flags |= O_EXCL;
+ // Set exclusive access if specified.
+ if ( options && options->flags & RTAUDIO_HOG_DEVICE ) flags |= O_EXCL;
- // Try to open the device.
- int fd;
- fd = open( ainfo.devnode, flags, 0 );
- if ( fd == -1 ) {
- if ( errno == EBUSY )
- errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") is busy.";
- else
- errorStream_ << "RtApiOss::probeDeviceOpen: error opening device (" << ainfo.name << ").";
- errorText_ = errorStream_.str();
- return FAILURE;
+ // Try to open the device.
+ int fd;
+ fd = open( ainfo.devnode, flags, 0 );
+ if ( fd == -1 ) {
+ if ( errno == EBUSY )
+ errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") is busy.";
+ else
+ errorStream_ << "RtApiOss::probeDeviceOpen: error opening device (" << ainfo.name << ").";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
+
+ // For duplex operation, specifically set this mode (this doesn't seem to work).
+ /*
+ if ( flags | O_RDWR ) {
+ result = ioctl( fd, SNDCTL_DSP_SETDUPLEX, NULL );
+ if ( result == -1) {
+ errorStream_ << "RtApiOss::probeDeviceOpen: error setting duplex mode for device (" << ainfo.name << ").";
+ errorText_ = errorStream_.str();
+ return FAILURE;
}
+ }
+ */
- // For duplex operation, specifically set this mode (this doesn't seem to work).
- /*
- if ( flags | O_RDWR ) {
- result = ioctl( fd, SNDCTL_DSP_SETDUPLEX, NULL );
- if ( result == -1) {
- errorStream_ << "RtApiOss::probeDeviceOpen: error setting duplex mode for device (" << ainfo.name << ").";
- errorText_ = errorStream_.str();
- return FAILURE;
- }
- }
- */
+ // Check the device channel support.
+ stream_.nUserChannels[mode] = channels;
+ if ( ainfo.max_channels < (int)(channels + firstChannel) ) {
+ close( fd );
+ errorStream_ << "RtApiOss::probeDeviceOpen: the device (" << ainfo.name << ") does not support requested channel parameters.";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
- // Check the device channel support.
- stream_.nUserChannels[mode] = channels;
- if ( ainfo.max_channels < (int)(channels + firstChannel) ) {
- close( fd );
- errorStream_ << "RtApiOss::probeDeviceOpen: the device (" << ainfo.name << ") does not support requested channel parameters.";
- errorText_ = errorStream_.str();
- return FAILURE;
- }
+ // Set the number of channels.
+ int deviceChannels = channels + firstChannel;
+ result = ioctl( fd, SNDCTL_DSP_CHANNELS, &deviceChannels );
+ if ( result == -1 || deviceChannels < (int)(channels + firstChannel) ) {
+ close( fd );
+ errorStream_ << "RtApiOss::probeDeviceOpen: error setting channel parameters on device (" << ainfo.name << ").";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
+ stream_.nDeviceChannels[mode] = deviceChannels;
- // Set the number of channels.
- int deviceChannels = channels + firstChannel;
- result = ioctl( fd, SNDCTL_DSP_CHANNELS, &deviceChannels );
- if ( result == -1 || deviceChannels < (int)(channels + firstChannel) ) {
- close( fd );
- errorStream_ << "RtApiOss::probeDeviceOpen: error setting channel parameters on device (" << ainfo.name << ").";
- errorText_ = errorStream_.str();
- return FAILURE;
- }
- stream_.nDeviceChannels[mode] = deviceChannels;
+ // Get the data format mask
+ int mask;
+ result = ioctl( fd, SNDCTL_DSP_GETFMTS, &mask );
+ if ( result == -1 ) {
+ close( fd );
+ errorStream_ << "RtApiOss::probeDeviceOpen: error getting device (" << ainfo.name << ") data formats.";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
- // Get the data format mask
- int mask;
- result = ioctl( fd, SNDCTL_DSP_GETFMTS, &mask );
- if ( result == -1 ) {
- close( fd );
- errorStream_ << "RtApiOss::probeDeviceOpen: error getting device (" << ainfo.name << ") data formats.";
- errorText_ = errorStream_.str();
- return FAILURE;
+ // Determine how to set the device format.
+ stream_.userFormat = format;
+ int deviceFormat = -1;
+ stream_.doByteSwap[mode] = false;
+ if ( format == RTAUDIO_SINT8 ) {
+ if ( mask & AFMT_S8 ) {
+ deviceFormat = AFMT_S8;
+ stream_.deviceFormat[mode] = RTAUDIO_SINT8;
}
-
- // Determine how to set the device format.
- stream_.userFormat = format;
- int deviceFormat = -1;
- stream_.doByteSwap[mode] = false;
- if ( format == RTAUDIO_SINT8 ) {
- if ( mask & AFMT_S8 ) {
- deviceFormat = AFMT_S8;
- stream_.deviceFormat[mode] = RTAUDIO_SINT8;
- }
+ }
+ else if ( format == RTAUDIO_SINT16 ) {
+ if ( mask & AFMT_S16_NE ) {
+ deviceFormat = AFMT_S16_NE;
+ stream_.deviceFormat[mode] = RTAUDIO_SINT16;
}
- else if ( format == RTAUDIO_SINT16 ) {
- if ( mask & AFMT_S16_NE ) {
- deviceFormat = AFMT_S16_NE;
- stream_.deviceFormat[mode] = RTAUDIO_SINT16;
- }
- else if ( mask & AFMT_S16_OE ) {
- deviceFormat = AFMT_S16_OE;
- stream_.deviceFormat[mode] = RTAUDIO_SINT16;
- stream_.doByteSwap[mode] = true;
- }
+ else if ( mask & AFMT_S16_OE ) {
+ deviceFormat = AFMT_S16_OE;
+ stream_.deviceFormat[mode] = RTAUDIO_SINT16;
+ stream_.doByteSwap[mode] = true;
}
- else if ( format == RTAUDIO_SINT24 ) {
- if ( mask & AFMT_S24_NE ) {
- deviceFormat = AFMT_S24_NE;
- stream_.deviceFormat[mode] = RTAUDIO_SINT24;
- }
- else if ( mask & AFMT_S24_OE ) {
- deviceFormat = AFMT_S24_OE;
- stream_.deviceFormat[mode] = RTAUDIO_SINT24;
- stream_.doByteSwap[mode] = true;
- }
+ }
+ else if ( format == RTAUDIO_SINT24 ) {
+ if ( mask & AFMT_S24_NE ) {
+ deviceFormat = AFMT_S24_NE;
+ stream_.deviceFormat[mode] = RTAUDIO_SINT24;
}
- else if ( format == RTAUDIO_SINT32 ) {
- if ( mask & AFMT_S32_NE ) {
- deviceFormat = AFMT_S32_NE;
- stream_.deviceFormat[mode] = RTAUDIO_SINT32;
- }
- else if ( mask & AFMT_S32_OE ) {
- deviceFormat = AFMT_S32_OE;
- stream_.deviceFormat[mode] = RTAUDIO_SINT32;
- stream_.doByteSwap[mode] = true;
- }
+ else if ( mask & AFMT_S24_OE ) {
+ deviceFormat = AFMT_S24_OE;
+ stream_.deviceFormat[mode] = RTAUDIO_SINT24;
+ stream_.doByteSwap[mode] = true;
}
-
- if ( deviceFormat == -1 ) {
- // The user requested format is not natively supported by the device.
- if ( mask & AFMT_S16_NE ) {
- deviceFormat = AFMT_S16_NE;
- stream_.deviceFormat[mode] = RTAUDIO_SINT16;
- }
- else if ( mask & AFMT_S32_NE ) {
- deviceFormat = AFMT_S32_NE;
- stream_.deviceFormat[mode] = RTAUDIO_SINT32;
- }
- else if ( mask & AFMT_S24_NE ) {
- deviceFormat = AFMT_S24_NE;
- stream_.deviceFormat[mode] = RTAUDIO_SINT24;
- }
- else if ( mask & AFMT_S16_OE ) {
- deviceFormat = AFMT_S16_OE;
- stream_.deviceFormat[mode] = RTAUDIO_SINT16;
- stream_.doByteSwap[mode] = true;
- }
- else if ( mask & AFMT_S32_OE ) {
- deviceFormat = AFMT_S32_OE;
- stream_.deviceFormat[mode] = RTAUDIO_SINT32;
- stream_.doByteSwap[mode] = true;
- }
- else if ( mask & AFMT_S24_OE ) {
- deviceFormat = AFMT_S24_OE;
- stream_.deviceFormat[mode] = RTAUDIO_SINT24;
- stream_.doByteSwap[mode] = true;
- }
- else if ( mask & AFMT_S8) {
- deviceFormat = AFMT_S8;
- stream_.deviceFormat[mode] = RTAUDIO_SINT8;
- }
+ }
+ else if ( format == RTAUDIO_SINT32 ) {
+ if ( mask & AFMT_S32_NE ) {
+ deviceFormat = AFMT_S32_NE;
+ stream_.deviceFormat[mode] = RTAUDIO_SINT32;
}
-
- if ( stream_.deviceFormat[mode] == 0 ) {
- // This really shouldn't happen ...
- close( fd );
- errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") data format not supported by RtAudio.";
- errorText_ = errorStream_.str();
- return FAILURE;
+ else if ( mask & AFMT_S32_OE ) {
+ deviceFormat = AFMT_S32_OE;
+ stream_.deviceFormat[mode] = RTAUDIO_SINT32;
+ stream_.doByteSwap[mode] = true;
}
+ }
- // Set the data format.
- int temp = deviceFormat;
- result = ioctl( fd, SNDCTL_DSP_SETFMT, &deviceFormat );
- if ( result == -1 || deviceFormat != temp ) {
- close( fd );
- errorStream_ << "RtApiOss::probeDeviceOpen: error setting data format on device (" << ainfo.name << ").";
- errorText_ = errorStream_.str();
- return FAILURE;
+ if ( deviceFormat == -1 ) {
+ // The user requested format is not natively supported by the device.
+ if ( mask & AFMT_S16_NE ) {
+ deviceFormat = AFMT_S16_NE;
+ stream_.deviceFormat[mode] = RTAUDIO_SINT16;
}
-
- // Attempt to set the buffer size. According to OSS, the minimum
- // number of buffers is two. The supposed minimum buffer size is 16
- // bytes, so that will be our lower bound. The argument to this
- // call is in the form 0xMMMMSSSS (hex), where the buffer size (in
- // bytes) is given as 2^SSSS and the number of buffers as 2^MMMM.
- // We'll check the actual value used near the end of the setup
- // procedure.
- int ossBufferBytes = *bufferSize * formatBytes( stream_.deviceFormat[mode] ) * deviceChannels;
- if ( ossBufferBytes < 16 ) ossBufferBytes = 16;
- int buffers = 0;
- if ( options ) buffers = options->numberOfBuffers;
- if ( options && options->flags & RTAUDIO_MINIMIZE_LATENCY ) buffers = 2;
- if ( buffers < 2 ) buffers = 3;
- temp = ((int) buffers << 16) + (int)( log10( (double)ossBufferBytes ) / log10( 2.0 ) );
- result = ioctl( fd, SNDCTL_DSP_SETFRAGMENT, &temp );
- if ( result == -1 ) {
- close( fd );
- errorStream_ << "RtApiOss::probeDeviceOpen: error setting buffer size on device (" << ainfo.name << ").";
- errorText_ = errorStream_.str();
- return FAILURE;
+ else if ( mask & AFMT_S32_NE ) {
+ deviceFormat = AFMT_S32_NE;
+ stream_.deviceFormat[mode] = RTAUDIO_SINT32;
+ }
+ else if ( mask & AFMT_S24_NE ) {
+ deviceFormat = AFMT_S24_NE;
+ stream_.deviceFormat[mode] = RTAUDIO_SINT24;
+ }
+ else if ( mask & AFMT_S16_OE ) {
+ deviceFormat = AFMT_S16_OE;
+ stream_.deviceFormat[mode] = RTAUDIO_SINT16;
+ stream_.doByteSwap[mode] = true;
+ }
+ else if ( mask & AFMT_S32_OE ) {
+ deviceFormat = AFMT_S32_OE;
+ stream_.deviceFormat[mode] = RTAUDIO_SINT32;
+ stream_.doByteSwap[mode] = true;
+ }
+ else if ( mask & AFMT_S24_OE ) {
+ deviceFormat = AFMT_S24_OE;
+ stream_.deviceFormat[mode] = RTAUDIO_SINT24;
+ stream_.doByteSwap[mode] = true;
+ }
+ else if ( mask & AFMT_S8) {
+ deviceFormat = AFMT_S8;
+ stream_.deviceFormat[mode] = RTAUDIO_SINT8;
}
- stream_.nBuffers = buffers;
+ }
- // Save buffer size (in sample frames).
- *bufferSize = ossBufferBytes / ( formatBytes(stream_.deviceFormat[mode]) * deviceChannels );
- stream_.bufferSize = *bufferSize;
+ if ( stream_.deviceFormat[mode] == 0 ) {
+ // This really shouldn't happen ...
+ close( fd );
+ errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") data format not supported by RtAudio.";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
- // Set the sample rate.
- int srate = sampleRate;
- result = ioctl( fd, SNDCTL_DSP_SPEED, &srate );
- if ( result == -1 ) {
- close( fd );
- errorStream_ << "RtApiOss::probeDeviceOpen: error setting sample rate (" << sampleRate << ") on device (" << ainfo.name << ").";
- errorText_ = errorStream_.str();
- return FAILURE;
- }
+ // Set the data format.
+ int temp = deviceFormat;
+ result = ioctl( fd, SNDCTL_DSP_SETFMT, &deviceFormat );
+ if ( result == -1 || deviceFormat != temp ) {
+ close( fd );
+ errorStream_ << "RtApiOss::probeDeviceOpen: error setting data format on device (" << ainfo.name << ").";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
- // Verify the sample rate setup worked.
- if ( abs( srate - sampleRate ) > 100 ) {
- close( fd );
- errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") does not support sample rate (" << sampleRate << ").";
- errorText_ = errorStream_.str();
- return FAILURE;
- }
- stream_.sampleRate = sampleRate;
+ // Attempt to set the buffer size. According to OSS, the minimum
+ // number of buffers is two. The supposed minimum buffer size is 16
+ // bytes, so that will be our lower bound. The argument to this
+ // call is in the form 0xMMMMSSSS (hex), where the buffer size (in
+ // bytes) is given as 2^SSSS and the number of buffers as 2^MMMM.
+ // We'll check the actual value used near the end of the setup
+ // procedure.
+ int ossBufferBytes = *bufferSize * formatBytes( stream_.deviceFormat[mode] ) * deviceChannels;
+ if ( ossBufferBytes < 16 ) ossBufferBytes = 16;
+ int buffers = 0;
+ if ( options ) buffers = options->numberOfBuffers;
+ if ( options && options->flags & RTAUDIO_MINIMIZE_LATENCY ) buffers = 2;
+ if ( buffers < 2 ) buffers = 3;
+ temp = ((int) buffers << 16) + (int)( log10( (double)ossBufferBytes ) / log10( 2.0 ) );
+ result = ioctl( fd, SNDCTL_DSP_SETFRAGMENT, &temp );
+ if ( result == -1 ) {
+ close( fd );
+ errorStream_ << "RtApiOss::probeDeviceOpen: error setting buffer size on device (" << ainfo.name << ").";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
+ stream_.nBuffers = buffers;
- if ( mode == INPUT && stream_.mode == OUTPUT && stream_.device[0] == device) {
- // We're doing duplex setup here.
- stream_.deviceFormat[0] = stream_.deviceFormat[1];
- stream_.nDeviceChannels[0] = deviceChannels;
- }
+ // Save buffer size (in sample frames).
+ *bufferSize = ossBufferBytes / ( formatBytes(stream_.deviceFormat[mode]) * deviceChannels );
+ stream_.bufferSize = *bufferSize;
- // Set interleaving parameters.
- stream_.userInterleaved = true;
- stream_.deviceInterleaved[mode] = true;
- if ( options && options->flags & RTAUDIO_NONINTERLEAVED )
- stream_.userInterleaved = false;
+ // Set the sample rate.
+ int srate = sampleRate;
+ result = ioctl( fd, SNDCTL_DSP_SPEED, &srate );
+ if ( result == -1 ) {
+ close( fd );
+ errorStream_ << "RtApiOss::probeDeviceOpen: error setting sample rate (" << sampleRate << ") on device (" << ainfo.name << ").";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
- // Set flags for buffer conversion
- stream_.doConvertBuffer[mode] = false;
- if ( stream_.userFormat != stream_.deviceFormat[mode] )
- stream_.doConvertBuffer[mode] = true;
- if ( stream_.nUserChannels[mode] < stream_.nDeviceChannels[mode] )
- stream_.doConvertBuffer[mode] = true;
- if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] &&
- stream_.nUserChannels[mode] > 1 )
- stream_.doConvertBuffer[mode] = true;
+ // Verify the sample rate setup worked.
+ if ( abs( srate - sampleRate ) > 100 ) {
+ close( fd );
+ errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") does not support sample rate (" << sampleRate << ").";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
+ stream_.sampleRate = sampleRate;
- // Allocate the stream handles if necessary and then save.
- if ( stream_.apiHandle == 0 ) {
- try {
- handle = new OssHandle;
- }
- catch ( std::bad_alloc& ) {
- errorText_ = "RtApiOss::probeDeviceOpen: error allocating OssHandle memory.";
- goto error;
- }
+ if ( mode == INPUT && stream_.mode == OUTPUT && stream_.device[0] == device) {
+ // We're doing duplex setup here.
+ stream_.deviceFormat[0] = stream_.deviceFormat[1];
+ stream_.nDeviceChannels[0] = deviceChannels;
+ }
- if ( pthread_cond_init( &handle->runnable, NULL ) ) {
- errorText_ = "RtApiOss::probeDeviceOpen: error initializing pthread condition variable.";
- goto error;
- }
+ // Set interleaving parameters.
+ stream_.userInterleaved = true;
+ stream_.deviceInterleaved[mode] = true;
+ if ( options && options->flags & RTAUDIO_NONINTERLEAVED )
+ stream_.userInterleaved = false;
+
+ // Set flags for buffer conversion
+ stream_.doConvertBuffer[mode] = false;
+ if ( stream_.userFormat != stream_.deviceFormat[mode] )
+ stream_.doConvertBuffer[mode] = true;
+ if ( stream_.nUserChannels[mode] < stream_.nDeviceChannels[mode] )
+ stream_.doConvertBuffer[mode] = true;
+ if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] &&
+ stream_.nUserChannels[mode] > 1 )
+ stream_.doConvertBuffer[mode] = true;
- stream_.apiHandle = (void *) handle;
+ // Allocate the stream handles if necessary and then save.
+ if ( stream_.apiHandle == 0 ) {
+ try {
+ handle = new OssHandle;
}
- else {
- handle = (OssHandle *) stream_.apiHandle;
+ catch ( std::bad_alloc& ) {
+ errorText_ = "RtApiOss::probeDeviceOpen: error allocating OssHandle memory.";
+ goto error;
}
- handle->id[mode] = fd;
- // Allocate necessary internal buffers.
- unsigned long bufferBytes;
- bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat );
- stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 );
- if ( stream_.userBuffer[mode] == NULL ) {
- errorText_ = "RtApiOss::probeDeviceOpen: error allocating user buffer memory.";
+ if ( pthread_cond_init( &handle->runnable, NULL ) ) {
+ errorText_ = "RtApiOss::probeDeviceOpen: error initializing pthread condition variable.";
goto error;
}
- if ( stream_.doConvertBuffer[mode] ) {
+ stream_.apiHandle = (void *) handle;
+ }
+ else {
+ handle = (OssHandle *) stream_.apiHandle;
+ }
+ handle->id[mode] = fd;
- bool makeBuffer = true;
- bufferBytes = stream_.nDeviceChannels[mode] * formatBytes( stream_.deviceFormat[mode] );
- if ( mode == INPUT ) {
- if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) {
- unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] );
- if ( bufferBytes <= bytesOut ) makeBuffer = false;
- }
+ // Allocate necessary internal buffers.
+ unsigned long bufferBytes;
+ bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat );
+ stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 );
+ if ( stream_.userBuffer[mode] == NULL ) {
+ errorText_ = "RtApiOss::probeDeviceOpen: error allocating user buffer memory.";
+ goto error;
+ }
+
+ if ( stream_.doConvertBuffer[mode] ) {
+
+ bool makeBuffer = true;
+ bufferBytes = stream_.nDeviceChannels[mode] * formatBytes( stream_.deviceFormat[mode] );
+ if ( mode == INPUT ) {
+ if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) {
+ unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] );
+ if ( bufferBytes <= bytesOut ) makeBuffer = false;
}
+ }
- if ( makeBuffer ) {
- bufferBytes *= *bufferSize;
- if ( stream_.deviceBuffer ) free( stream_.deviceBuffer );
- stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 );
- if ( stream_.deviceBuffer == NULL ) {
- errorText_ = "RtApiOss::probeDeviceOpen: error allocating device buffer memory.";
- goto error;
- }
+ if ( makeBuffer ) {
+ bufferBytes *= *bufferSize;
+ if ( stream_.deviceBuffer ) free( stream_.deviceBuffer );
+ stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 );
+ if ( stream_.deviceBuffer == NULL ) {
+ errorText_ = "RtApiOss::probeDeviceOpen: error allocating device buffer memory.";
+ goto error;
}
}
+ }
- stream_.device[mode] = device;
- stream_.state = STREAM_STOPPED;
+ stream_.device[mode] = device;
+ stream_.state = STREAM_STOPPED;
- // Setup the buffer conversion information structure.
- if ( stream_.doConvertBuffer[mode] ) setConvertInfo( mode, firstChannel );
+ // Setup the buffer conversion information structure.
+ if ( stream_.doConvertBuffer[mode] ) setConvertInfo( mode, firstChannel );
- // Setup thread if necessary.
- if ( stream_.mode == OUTPUT && mode == INPUT ) {
- // We had already set up an output stream.
- stream_.mode = DUPLEX;
- if ( stream_.device[0] == device ) handle->id[0] = fd;
- }
- else {
- stream_.mode = mode;
+ // Setup thread if necessary.
+ if ( stream_.mode == OUTPUT && mode == INPUT ) {
+ // We had already set up an output stream.
+ stream_.mode = DUPLEX;
+ if ( stream_.device[0] == device ) handle->id[0] = fd;
+ }
+ else {
+ stream_.mode = mode;
- // Setup callback thread.
- stream_.callbackInfo.object = (void *) this;
+ // Setup callback thread.
+ stream_.callbackInfo.object = (void *) this;
- // Set the thread attributes for joinable and realtime scheduling
- // priority. The higher priority will only take affect if the
- // program is run as root or suid.
- pthread_attr_t attr;
- pthread_attr_init( &attr );
- pthread_attr_setdetachstate( &attr, PTHREAD_CREATE_JOINABLE );
+ // Set the thread attributes for joinable and realtime scheduling
+ // priority. The higher priority will only take affect if the
+ // program is run as root or suid.
+ pthread_attr_t attr;
+ pthread_attr_init( &attr );
+ pthread_attr_setdetachstate( &attr, PTHREAD_CREATE_JOINABLE );
#ifdef SCHED_RR // Undefined with some OSes (eg: NetBSD 1.6.x with GNU Pthread)
- if ( options && options->flags & RTAUDIO_SCHEDULE_REALTIME ) {
- struct sched_param param;
- int priority = options->priority;
- int min = sched_get_priority_min( SCHED_RR );
- int max = sched_get_priority_max( SCHED_RR );
- if ( priority < min ) priority = min;
- else if ( priority > max ) priority = max;
- param.sched_priority = priority;
- pthread_attr_setschedparam( &attr, ¶m );
- pthread_attr_setschedpolicy( &attr, SCHED_RR );
- }
- else
- pthread_attr_setschedpolicy( &attr, SCHED_OTHER );
-#else
+ if ( options && options->flags & RTAUDIO_SCHEDULE_REALTIME ) {
+ struct sched_param param;
+ int priority = options->priority;
+ int min = sched_get_priority_min( SCHED_RR );
+ int max = sched_get_priority_max( SCHED_RR );
+ if ( priority < min ) priority = min;
+ else if ( priority > max ) priority = max;
+ param.sched_priority = priority;
+ pthread_attr_setschedparam( &attr, ¶m );
+ pthread_attr_setschedpolicy( &attr, SCHED_RR );
+ }
+ else
pthread_attr_setschedpolicy( &attr, SCHED_OTHER );
+#else
+ pthread_attr_setschedpolicy( &attr, SCHED_OTHER );
#endif
- stream_.callbackInfo.isRunning = true;
- result = pthread_create( &stream_.callbackInfo.thread, &attr, ossCallbackHandler, &stream_.callbackInfo );
- pthread_attr_destroy( &attr );
- if ( result ) {
- stream_.callbackInfo.isRunning = false;
- errorText_ = "RtApiOss::error creating callback thread!";
- goto error;
- }
+ stream_.callbackInfo.isRunning = true;
+ result = pthread_create( &stream_.callbackInfo.thread, &attr, ossCallbackHandler, &stream_.callbackInfo );
+ pthread_attr_destroy( &attr );
+ if ( result ) {
+ stream_.callbackInfo.isRunning = false;
+ errorText_ = "RtApiOss::error creating callback thread!";
+ goto error;
}
+ }
- return SUCCESS;
-
- error:
- if ( handle ) {
- pthread_cond_destroy( &handle->runnable );
- if ( handle->id[0] ) close( handle->id[0] );
- if ( handle->id[1] ) close( handle->id[1] );
- delete handle;
- stream_.apiHandle = 0;
- }
+ return SUCCESS;
- for ( int i=0; i<2; i++ ) {
- if ( stream_.userBuffer[i] ) {
- free( stream_.userBuffer[i] );
- stream_.userBuffer[i] = 0;
- }
- }
+ error:
+ if ( handle ) {
+ pthread_cond_destroy( &handle->runnable );
+ if ( handle->id[0] ) close( handle->id[0] );
+ if ( handle->id[1] ) close( handle->id[1] );
+ delete handle;
+ stream_.apiHandle = 0;
+ }
- if ( stream_.deviceBuffer ) {
- free( stream_.deviceBuffer );
- stream_.deviceBuffer = 0;
+ for ( int i=0; i<2; i++ ) {
+ if ( stream_.userBuffer[i] ) {
+ free( stream_.userBuffer[i] );
+ stream_.userBuffer[i] = 0;
}
+ }
- return FAILURE;
+ if ( stream_.deviceBuffer ) {
+ free( stream_.deviceBuffer );
+ stream_.deviceBuffer = 0;
}
- void RtApiOss :: closeStream()
- {
- if ( stream_.state == STREAM_CLOSED ) {
- errorText_ = "RtApiOss::closeStream(): no open stream to close!";
- error( RtError::WARNING );
- return;
- }
+ return FAILURE;
+}
- OssHandle *handle = (OssHandle *) stream_.apiHandle;
- stream_.callbackInfo.isRunning = false;
- MUTEX_LOCK( &stream_.mutex );
- if ( stream_.state == STREAM_STOPPED )
- pthread_cond_signal( &handle->runnable );
- MUTEX_UNLOCK( &stream_.mutex );
- pthread_join( stream_.callbackInfo.thread, NULL );
+void RtApiOss :: closeStream()
+{
+ if ( stream_.state == STREAM_CLOSED ) {
+ errorText_ = "RtApiOss::closeStream(): no open stream to close!";
+ error( RtError::WARNING );
+ return;
+ }
- if ( stream_.state == STREAM_RUNNING ) {
- if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX )
- ioctl( handle->id[0], SNDCTL_DSP_HALT, 0 );
- else
- ioctl( handle->id[1], SNDCTL_DSP_HALT, 0 );
- stream_.state = STREAM_STOPPED;
- }
+ OssHandle *handle = (OssHandle *) stream_.apiHandle;
+ stream_.callbackInfo.isRunning = false;
+ MUTEX_LOCK( &stream_.mutex );
+ if ( stream_.state == STREAM_STOPPED )
+ pthread_cond_signal( &handle->runnable );
+ MUTEX_UNLOCK( &stream_.mutex );
+ pthread_join( stream_.callbackInfo.thread, NULL );
- if ( handle ) {
- pthread_cond_destroy( &handle->runnable );
- if ( handle->id[0] ) close( handle->id[0] );
- if ( handle->id[1] ) close( handle->id[1] );
- delete handle;
- stream_.apiHandle = 0;
- }
+ if ( stream_.state == STREAM_RUNNING ) {
+ if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX )
+ ioctl( handle->id[0], SNDCTL_DSP_HALT, 0 );
+ else
+ ioctl( handle->id[1], SNDCTL_DSP_HALT, 0 );
+ stream_.state = STREAM_STOPPED;
+ }
- for ( int i=0; i<2; i++ ) {
- if ( stream_.userBuffer[i] ) {
- free( stream_.userBuffer[i] );
- stream_.userBuffer[i] = 0;
- }
- }
+ if ( handle ) {
+ pthread_cond_destroy( &handle->runnable );
+ if ( handle->id[0] ) close( handle->id[0] );
+ if ( handle->id[1] ) close( handle->id[1] );
+ delete handle;
+ stream_.apiHandle = 0;
+ }
- if ( stream_.deviceBuffer ) {
- free( stream_.deviceBuffer );
- stream_.deviceBuffer = 0;
+ for ( int i=0; i<2; i++ ) {
+ if ( stream_.userBuffer[i] ) {
+ free( stream_.userBuffer[i] );
+ stream_.userBuffer[i] = 0;
}
+ }
- stream_.mode = UNINITIALIZED;
- stream_.state = STREAM_CLOSED;
+ if ( stream_.deviceBuffer ) {
+ free( stream_.deviceBuffer );
+ stream_.deviceBuffer = 0;
}
- void RtApiOss :: startStream()
- {
- verifyStream();
- if ( stream_.state == STREAM_RUNNING ) {
- errorText_ = "RtApiOss::startStream(): the stream is already running!";
- error( RtError::WARNING );
- return;
- }
+ stream_.mode = UNINITIALIZED;
+ stream_.state = STREAM_CLOSED;
+}
- MUTEX_LOCK( &stream_.mutex );
+void RtApiOss :: startStream()
+{
+ verifyStream();
+ if ( stream_.state == STREAM_RUNNING ) {
+ errorText_ = "RtApiOss::startStream(): the stream is already running!";
+ error( RtError::WARNING );
+ return;
+ }
- stream_.state = STREAM_RUNNING;
+ MUTEX_LOCK( &stream_.mutex );
- // No need to do anything else here ... OSS automatically starts
- // when fed samples.
+ stream_.state = STREAM_RUNNING;
- MUTEX_UNLOCK( &stream_.mutex );
+ // No need to do anything else here ... OSS automatically starts
+ // when fed samples.
- OssHandle *handle = (OssHandle *) stream_.apiHandle;
- pthread_cond_signal( &handle->runnable );
- }
+ MUTEX_UNLOCK( &stream_.mutex );
- void RtApiOss :: stopStream()
- {
- verifyStream();
- if ( stream_.state == STREAM_STOPPED ) {
- errorText_ = "RtApiOss::stopStream(): the stream is already stopped!";
- error( RtError::WARNING );
- return;
- }
+ OssHandle *handle = (OssHandle *) stream_.apiHandle;
+ pthread_cond_signal( &handle->runnable );
+}
- // Change the state before the lock to improve shutdown response
- // when using a callback.
- stream_.state = STREAM_STOPPED;
- MUTEX_LOCK( &stream_.mutex );
+void RtApiOss :: stopStream()
+{
+ verifyStream();
+ if ( stream_.state == STREAM_STOPPED ) {
+ errorText_ = "RtApiOss::stopStream(): the stream is already stopped!";
+ error( RtError::WARNING );
+ return;
+ }
- int result = 0;
- OssHandle *handle = (OssHandle *) stream_.apiHandle;
- if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
+ MUTEX_LOCK( &stream_.mutex );
- // Flush the output with zeros a few times.
- char *buffer;
- int samples;
- RtAudioFormat format;
+ // The state might change while waiting on a mutex.
+ if ( stream_.state == STREAM_STOPPED ) {
+ MUTEX_UNLOCK( &stream_.mutex );
+ return;
+ }
- if ( stream_.doConvertBuffer[0] ) {
- buffer = stream_.deviceBuffer;
- samples = stream_.bufferSize * stream_.nDeviceChannels[0];
- format = stream_.deviceFormat[0];
- }
- else {
- buffer = stream_.userBuffer[0];
- samples = stream_.bufferSize * stream_.nUserChannels[0];
- format = stream_.userFormat;
- }
+ int result = 0;
+ OssHandle *handle = (OssHandle *) stream_.apiHandle;
+ if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
- memset( buffer, 0, samples * formatBytes(format) );
- for ( unsigned int i=0; i<stream_.nBuffers+1; i++ ) {
- result = write( handle->id[0], buffer, samples * formatBytes(format) );
- if ( result == -1 ) {
- errorText_ = "RtApiOss::stopStream: audio write error.";
- error( RtError::WARNING );
- }
- }
+ // Flush the output with zeros a few times.
+ char *buffer;
+ int samples;
+ RtAudioFormat format;
- result = ioctl( handle->id[0], SNDCTL_DSP_HALT, 0 );
- if ( result == -1 ) {
- errorStream_ << "RtApiOss::stopStream: system error stopping callback procedure on device (" << stream_.device[0] << ").";
- errorText_ = errorStream_.str();
- goto unlock;
- }
- handle->triggered = false;
+ if ( stream_.doConvertBuffer[0] ) {
+ buffer = stream_.deviceBuffer;
+ samples = stream_.bufferSize * stream_.nDeviceChannels[0];
+ format = stream_.deviceFormat[0];
+ }
+ else {
+ buffer = stream_.userBuffer[0];
+ samples = stream_.bufferSize * stream_.nUserChannels[0];
+ format = stream_.userFormat;
}
- if ( stream_.mode == INPUT || ( stream_.mode == DUPLEX && handle->id[0] != handle->id[1] ) ) {
- result = ioctl( handle->id[1], SNDCTL_DSP_HALT, 0 );
+ memset( buffer, 0, samples * formatBytes(format) );
+ for ( unsigned int i=0; i<stream_.nBuffers+1; i++ ) {
+ result = write( handle->id[0], buffer, samples * formatBytes(format) );
if ( result == -1 ) {
- errorStream_ << "RtApiOss::stopStream: system error stopping input callback procedure on device (" << stream_.device[0] << ").";
- errorText_ = errorStream_.str();
- goto unlock;
+ errorText_ = "RtApiOss::stopStream: audio write error.";
+ error( RtError::WARNING );
}
}
- unlock:
- MUTEX_UNLOCK( &stream_.mutex );
-
- stream_.state = STREAM_STOPPED;
- if ( result != -1 ) return;
- error( RtError::SYSTEM_ERROR );
+ result = ioctl( handle->id[0], SNDCTL_DSP_HALT, 0 );
+ if ( result == -1 ) {
+ errorStream_ << "RtApiOss::stopStream: system error stopping callback procedure on device (" << stream_.device[0] << ").";
+ errorText_ = errorStream_.str();
+ goto unlock;
+ }
+ handle->triggered = false;
}
- void RtApiOss :: abortStream()
- {
- verifyStream();
- if ( stream_.state == STREAM_STOPPED ) {
- errorText_ = "RtApiOss::abortStream(): the stream is already stopped!";
- error( RtError::WARNING );
- return;
+ if ( stream_.mode == INPUT || ( stream_.mode == DUPLEX && handle->id[0] != handle->id[1] ) ) {
+ result = ioctl( handle->id[1], SNDCTL_DSP_HALT, 0 );
+ if ( result == -1 ) {
+ errorStream_ << "RtApiOss::stopStream: system error stopping input callback procedure on device (" << stream_.device[0] << ").";
+ errorText_ = errorStream_.str();
+ goto unlock;
}
+ }
- // Change the state before the lock to improve shutdown response
- // when using a callback.
- stream_.state = STREAM_STOPPED;
- MUTEX_LOCK( &stream_.mutex );
+ unlock:
+ stream_.state = STREAM_STOPPED;
+ MUTEX_UNLOCK( &stream_.mutex );
- int result = 0;
- OssHandle *handle = (OssHandle *) stream_.apiHandle;
- if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
- result = ioctl( handle->id[0], SNDCTL_DSP_HALT, 0 );
- if ( result == -1 ) {
- errorStream_ << "RtApiOss::abortStream: system error stopping callback procedure on device (" << stream_.device[0] << ").";
- errorText_ = errorStream_.str();
- goto unlock;
- }
- handle->triggered = false;
- }
+ if ( result != -1 ) return;
+ error( RtError::SYSTEM_ERROR );
+}
- if ( stream_.mode == INPUT || ( stream_.mode == DUPLEX && handle->id[0] != handle->id[1] ) ) {
- result = ioctl( handle->id[1], SNDCTL_DSP_HALT, 0 );
- if ( result == -1 ) {
- errorStream_ << "RtApiOss::abortStream: system error stopping input callback procedure on device (" << stream_.device[0] << ").";
- errorText_ = errorStream_.str();
- goto unlock;
- }
- }
+void RtApiOss :: abortStream()
+{
+ verifyStream();
+ if ( stream_.state == STREAM_STOPPED ) {
+ errorText_ = "RtApiOss::abortStream(): the stream is already stopped!";
+ error( RtError::WARNING );
+ return;
+ }
+
+ MUTEX_LOCK( &stream_.mutex );
- unlock:
+ // The state might change while waiting on a mutex.
+ if ( stream_.state == STREAM_STOPPED ) {
MUTEX_UNLOCK( &stream_.mutex );
-
- stream_.state = STREAM_STOPPED;
- if ( result != -1 ) return;
- error( RtError::SYSTEM_ERROR );
+ return;
}
- void RtApiOss :: callbackEvent()
- {
- OssHandle *handle = (OssHandle *) stream_.apiHandle;
- if ( stream_.state == STREAM_STOPPED ) {
- MUTEX_LOCK( &stream_.mutex );
- pthread_cond_wait( &handle->runnable, &stream_.mutex );
- if ( stream_.state != STREAM_RUNNING ) {
- MUTEX_UNLOCK( &stream_.mutex );
- return;
- }
- MUTEX_UNLOCK( &stream_.mutex );
+ int result = 0;
+ OssHandle *handle = (OssHandle *) stream_.apiHandle;
+ if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
+ result = ioctl( handle->id[0], SNDCTL_DSP_HALT, 0 );
+ if ( result == -1 ) {
+ errorStream_ << "RtApiOss::abortStream: system error stopping callback procedure on device (" << stream_.device[0] << ").";
+ errorText_ = errorStream_.str();
+ goto unlock;
}
+ handle->triggered = false;
+ }
- if ( stream_.state == STREAM_CLOSED ) {
- errorText_ = "RtApiOss::callbackEvent(): the stream is closed ... this shouldn't happen!";
- error( RtError::WARNING );
- return;
+ if ( stream_.mode == INPUT || ( stream_.mode == DUPLEX && handle->id[0] != handle->id[1] ) ) {
+ result = ioctl( handle->id[1], SNDCTL_DSP_HALT, 0 );
+ if ( result == -1 ) {
+ errorStream_ << "RtApiOss::abortStream: system error stopping input callback procedure on device (" << stream_.device[0] << ").";
+ errorText_ = errorStream_.str();
+ goto unlock;
}
+ }
- // Invoke user callback to get fresh output data.
- int doStopStream = 0;
- RtAudioCallback callback = (RtAudioCallback) stream_.callbackInfo.callback;
- double streamTime = getStreamTime();
- RtAudioStreamStatus status = 0;
- if ( stream_.mode != INPUT && handle->xrun[0] == true ) {
- status |= RTAUDIO_OUTPUT_UNDERFLOW;
- handle->xrun[0] = false;
- }
- if ( stream_.mode != OUTPUT && handle->xrun[1] == true ) {
- status |= RTAUDIO_INPUT_OVERFLOW;
- handle->xrun[1] = false;
- }
- doStopStream = callback( stream_.userBuffer[0], stream_.userBuffer[1],
- stream_.bufferSize, streamTime, status, stream_.callbackInfo.userData );
- if ( doStopStream == 2 ) {
- this->abortStream();
+ unlock:
+ stream_.state = STREAM_STOPPED;
+ MUTEX_UNLOCK( &stream_.mutex );
+
+ if ( result != -1 ) return;
+ error( RtError::SYSTEM_ERROR );
+}
+
+void RtApiOss :: callbackEvent()
+{
+ OssHandle *handle = (OssHandle *) stream_.apiHandle;
+ if ( stream_.state == STREAM_STOPPED ) {
+ MUTEX_LOCK( &stream_.mutex );
+ pthread_cond_wait( &handle->runnable, &stream_.mutex );
+ if ( stream_.state != STREAM_RUNNING ) {
+ MUTEX_UNLOCK( &stream_.mutex );
return;
}
+ MUTEX_UNLOCK( &stream_.mutex );
+ }
- MUTEX_LOCK( &stream_.mutex );
+ if ( stream_.state == STREAM_CLOSED ) {
+ errorText_ = "RtApiOss::callbackEvent(): the stream is closed ... this shouldn't happen!";
+ error( RtError::WARNING );
+ return;
+ }
- // The state might change while waiting on a mutex.
- if ( stream_.state == STREAM_STOPPED ) goto unlock;
+ // Invoke user callback to get fresh output data.
+ int doStopStream = 0;
+ RtAudioCallback callback = (RtAudioCallback) stream_.callbackInfo.callback;
+ double streamTime = getStreamTime();
+ RtAudioStreamStatus status = 0;
+ if ( stream_.mode != INPUT && handle->xrun[0] == true ) {
+ status |= RTAUDIO_OUTPUT_UNDERFLOW;
+ handle->xrun[0] = false;
+ }
+ if ( stream_.mode != OUTPUT && handle->xrun[1] == true ) {
+ status |= RTAUDIO_INPUT_OVERFLOW;
+ handle->xrun[1] = false;
+ }
+ doStopStream = callback( stream_.userBuffer[0], stream_.userBuffer[1],
+ stream_.bufferSize, streamTime, status, stream_.callbackInfo.userData );
+ if ( doStopStream == 2 ) {
+ this->abortStream();
+ return;
+ }
- int result;
- char *buffer;
- int samples;
- RtAudioFormat format;
+ MUTEX_LOCK( &stream_.mutex );
- if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
+ // The state might change while waiting on a mutex.
+ if ( stream_.state == STREAM_STOPPED ) goto unlock;
- // Setup parameters and do buffer conversion if necessary.
- if ( stream_.doConvertBuffer[0] ) {
- buffer = stream_.deviceBuffer;
- convertBuffer( buffer, stream_.userBuffer[0], stream_.convertInfo[0] );
- samples = stream_.bufferSize * stream_.nDeviceChannels[0];
- format = stream_.deviceFormat[0];
- }
- else {
- buffer = stream_.userBuffer[0];
- samples = stream_.bufferSize * stream_.nUserChannels[0];
- format = stream_.userFormat;
- }
+ int result;
+ char *buffer;
+ int samples;
+ RtAudioFormat format;
- // Do byte swapping if necessary.
- if ( stream_.doByteSwap[0] )
- byteSwapBuffer( buffer, samples, format );
-
- if ( stream_.mode == DUPLEX && handle->triggered == false ) {
- int trig = 0;
- ioctl( handle->id[0], SNDCTL_DSP_SETTRIGGER, &trig );
- result = write( handle->id[0], buffer, samples * formatBytes(format) );
- trig = PCM_ENABLE_INPUT|PCM_ENABLE_OUTPUT;
- ioctl( handle->id[0], SNDCTL_DSP_SETTRIGGER, &trig );
- handle->triggered = true;
- }
- else
- // Write samples to device.
- result = write( handle->id[0], buffer, samples * formatBytes(format) );
+ if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
- if ( result == -1 ) {
- // We'll assume this is an underrun, though there isn't a
- // specific means for determining that.
- handle->xrun[0] = true;
- errorText_ = "RtApiOss::callbackEvent: audio write error.";
- error( RtError::WARNING );
- // Continue on to input section.
- }
+ // Setup parameters and do buffer conversion if necessary.
+ if ( stream_.doConvertBuffer[0] ) {
+ buffer = stream_.deviceBuffer;
+ convertBuffer( buffer, stream_.userBuffer[0], stream_.convertInfo[0] );
+ samples = stream_.bufferSize * stream_.nDeviceChannels[0];
+ format = stream_.deviceFormat[0];
+ }
+ else {
+ buffer = stream_.userBuffer[0];
+ samples = stream_.bufferSize * stream_.nUserChannels[0];
+ format = stream_.userFormat;
}
- if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {
+ // Do byte swapping if necessary.
+ if ( stream_.doByteSwap[0] )
+ byteSwapBuffer( buffer, samples, format );
- // Setup parameters.
- if ( stream_.doConvertBuffer[1] ) {
- buffer = stream_.deviceBuffer;
- samples = stream_.bufferSize * stream_.nDeviceChannels[1];
- format = stream_.deviceFormat[1];
- }
- else {
- buffer = stream_.userBuffer[1];
- samples = stream_.bufferSize * stream_.nUserChannels[1];
- format = stream_.userFormat;
- }
+ if ( stream_.mode == DUPLEX && handle->triggered == false ) {
+ int trig = 0;
+ ioctl( handle->id[0], SNDCTL_DSP_SETTRIGGER, &trig );
+ result = write( handle->id[0], buffer, samples * formatBytes(format) );
+ trig = PCM_ENABLE_INPUT|PCM_ENABLE_OUTPUT;
+ ioctl( handle->id[0], SNDCTL_DSP_SETTRIGGER, &trig );
+ handle->triggered = true;
+ }
+ else
+ // Write samples to device.
+ result = write( handle->id[0], buffer, samples * formatBytes(format) );
+
+ if ( result == -1 ) {
+ // We'll assume this is an underrun, though there isn't a
+ // specific means for determining that.
+ handle->xrun[0] = true;
+ errorText_ = "RtApiOss::callbackEvent: audio write error.";
+ error( RtError::WARNING );
+ // Continue on to input section.
+ }
+ }
- // Read samples from device.
- result = read( handle->id[1], buffer, samples * formatBytes(format) );
+ if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {
- if ( result == -1 ) {
- // We'll assume this is an overrun, though there isn't a
- // specific means for determining that.
- handle->xrun[1] = true;
- errorText_ = "RtApiOss::callbackEvent: audio read error.";
- error( RtError::WARNING );
- goto unlock;
- }
+ // Setup parameters.
+ if ( stream_.doConvertBuffer[1] ) {
+ buffer = stream_.deviceBuffer;
+ samples = stream_.bufferSize * stream_.nDeviceChannels[1];
+ format = stream_.deviceFormat[1];
+ }
+ else {
+ buffer = stream_.userBuffer[1];
+ samples = stream_.bufferSize * stream_.nUserChannels[1];
+ format = stream_.userFormat;
+ }
- // Do byte swapping if necessary.
- if ( stream_.doByteSwap[1] )
- byteSwapBuffer( buffer, samples, format );
+ // Read samples from device.
+ result = read( handle->id[1], buffer, samples * formatBytes(format) );
- // Do buffer conversion if necessary.
- if ( stream_.doConvertBuffer[1] )
- convertBuffer( stream_.userBuffer[1], stream_.deviceBuffer, stream_.convertInfo[1] );
+ if ( result == -1 ) {
+ // We'll assume this is an overrun, though there isn't a
+ // specific means for determining that.
+ handle->xrun[1] = true;
+ errorText_ = "RtApiOss::callbackEvent: audio read error.";
+ error( RtError::WARNING );
+ goto unlock;
}
- unlock:
- MUTEX_UNLOCK( &stream_.mutex );
+ // Do byte swapping if necessary.
+ if ( stream_.doByteSwap[1] )
+ byteSwapBuffer( buffer, samples, format );
- RtApi::tickStreamTime();
- if ( doStopStream == 1 ) this->stopStream();
+ // Do buffer conversion if necessary.
+ if ( stream_.doConvertBuffer[1] )
+ convertBuffer( stream_.userBuffer[1], stream_.deviceBuffer, stream_.convertInfo[1] );
}
- extern "C" void *ossCallbackHandler( void *ptr )
- {
- CallbackInfo *info = (CallbackInfo *) ptr;
- RtApiOss *object = (RtApiOss *) info->object;
- bool *isRunning = &info->isRunning;
+ unlock:
+ MUTEX_UNLOCK( &stream_.mutex );
- while ( *isRunning == true ) {
- pthread_testcancel();
- object->callbackEvent();
- }
+ RtApi::tickStreamTime();
+ if ( doStopStream == 1 ) this->stopStream();
+}
+
+extern "C" void *ossCallbackHandler( void *ptr )
+{
+ CallbackInfo *info = (CallbackInfo *) ptr;
+ RtApiOss *object = (RtApiOss *) info->object;
+ bool *isRunning = &info->isRunning;
- pthread_exit( NULL );
+ while ( *isRunning == true ) {
+ pthread_testcancel();
+ object->callbackEvent();
}
- //******************** End of __LINUX_OSS__ *********************//
+ pthread_exit( NULL );
+}
+
+//******************** End of __LINUX_OSS__ *********************//
#endif
- // *************************************************** //
- //
- // Protected common (OS-independent) RtAudio methods.
- //
- // *************************************************** //
+// *************************************************** //
+//
+// Protected common (OS-independent) RtAudio methods.
+//
+// *************************************************** //
- // This method can be modified to control the behavior of error
- // message printing.
- void RtApi :: error( RtError::Type type )
- {
- errorStream_.str(""); // clear the ostringstream
- if ( type == RtError::WARNING && showWarnings_ == true )
- std::cerr << '\n' << errorText_ << "\n\n";
- else
- throw( RtError( errorText_, type ) );
- }
+// This method can be modified to control the behavior of error
+// message printing.
+void RtApi :: error( RtError::Type type )
+{
+ errorStream_.str(""); // clear the ostringstream
+ if ( type == RtError::WARNING && showWarnings_ == true )
+ std::cerr << '\n' << errorText_ << "\n\n";
+ else
+ throw( RtError( errorText_, type ) );
+}
- void RtApi :: verifyStream()
- {
- if ( stream_.state == STREAM_CLOSED ) {
- errorText_ = "RtApi:: a stream is not open!";
- error( RtError::INVALID_USE );
- }
+void RtApi :: verifyStream()
+{
+ if ( stream_.state == STREAM_CLOSED ) {
+ errorText_ = "RtApi:: a stream is not open!";
+ error( RtError::INVALID_USE );
}
+}
- void RtApi :: clearStreamInfo()
- {
- stream_.mode = UNINITIALIZED;
- stream_.state = STREAM_CLOSED;
- stream_.sampleRate = 0;
- stream_.bufferSize = 0;
- stream_.nBuffers = 0;
- stream_.userFormat = 0;
- stream_.userInterleaved = true;
- stream_.streamTime = 0.0;
- stream_.apiHandle = 0;
- stream_.deviceBuffer = 0;
- stream_.callbackInfo.callback = 0;
- stream_.callbackInfo.userData = 0;
- stream_.callbackInfo.isRunning = false;
- for ( int i=0; i<2; i++ ) {
- stream_.device[i] = 11111;
- stream_.doConvertBuffer[i] = false;
- stream_.deviceInterleaved[i] = true;
- stream_.doByteSwap[i] = false;
- stream_.nUserChannels[i] = 0;
- stream_.nDeviceChannels[i] = 0;
- stream_.channelOffset[i] = 0;
- stream_.deviceFormat[i] = 0;
- stream_.latency[i] = 0;
- stream_.userBuffer[i] = 0;
- stream_.convertInfo[i].channels = 0;
- stream_.convertInfo[i].inJump = 0;
- stream_.convertInfo[i].outJump = 0;
- stream_.convertInfo[i].inFormat = 0;
- stream_.convertInfo[i].outFormat = 0;
- stream_.convertInfo[i].inOffset.clear();
- stream_.convertInfo[i].outOffset.clear();
- }
+void RtApi :: clearStreamInfo()
+{
+ stream_.mode = UNINITIALIZED;
+ stream_.state = STREAM_CLOSED;
+ stream_.sampleRate = 0;
+ stream_.bufferSize = 0;
+ stream_.nBuffers = 0;
+ stream_.userFormat = 0;
+ stream_.userInterleaved = true;
+ stream_.streamTime = 0.0;
+ stream_.apiHandle = 0;
+ stream_.deviceBuffer = 0;
+ stream_.callbackInfo.callback = 0;
+ stream_.callbackInfo.userData = 0;
+ stream_.callbackInfo.isRunning = false;
+ for ( int i=0; i<2; i++ ) {
+ stream_.device[i] = 11111;
+ stream_.doConvertBuffer[i] = false;
+ stream_.deviceInterleaved[i] = true;
+ stream_.doByteSwap[i] = false;
+ stream_.nUserChannels[i] = 0;
+ stream_.nDeviceChannels[i] = 0;
+ stream_.channelOffset[i] = 0;
+ stream_.deviceFormat[i] = 0;
+ stream_.latency[i] = 0;
+ stream_.userBuffer[i] = 0;
+ stream_.convertInfo[i].channels = 0;
+ stream_.convertInfo[i].inJump = 0;
+ stream_.convertInfo[i].outJump = 0;
+ stream_.convertInfo[i].inFormat = 0;
+ stream_.convertInfo[i].outFormat = 0;
+ stream_.convertInfo[i].inOffset.clear();
+ stream_.convertInfo[i].outOffset.clear();
}
+}
- unsigned int RtApi :: formatBytes( RtAudioFormat format )
- {
- if ( format == RTAUDIO_SINT16 )
- return 2;
- else if ( format == RTAUDIO_SINT24 || format == RTAUDIO_SINT32 ||
- format == RTAUDIO_FLOAT32 )
- return 4;
- else if ( format == RTAUDIO_FLOAT64 )
- return 8;
- else if ( format == RTAUDIO_SINT8 )
- return 1;
-
- errorText_ = "RtApi::formatBytes: undefined format.";
- error( RtError::WARNING );
+unsigned int RtApi :: formatBytes( RtAudioFormat format )
+{
+ if ( format == RTAUDIO_SINT16 )
+ return 2;
+ else if ( format == RTAUDIO_SINT24 || format == RTAUDIO_SINT32 ||
+ format == RTAUDIO_FLOAT32 )
+ return 4;
+ else if ( format == RTAUDIO_FLOAT64 )
+ return 8;
+ else if ( format == RTAUDIO_SINT8 )
+ return 1;
+
+ errorText_ = "RtApi::formatBytes: undefined format.";
+ error( RtError::WARNING );
- return 0;
+ return 0;
+}
+
+void RtApi :: setConvertInfo( StreamMode mode, unsigned int firstChannel )
+{
+ if ( mode == INPUT ) { // convert device to user buffer
+ stream_.convertInfo[mode].inJump = stream_.nDeviceChannels[1];
+ stream_.convertInfo[mode].outJump = stream_.nUserChannels[1];
+ stream_.convertInfo[mode].inFormat = stream_.deviceFormat[1];
+ stream_.convertInfo[mode].outFormat = stream_.userFormat;
+ }
+ else { // convert user to device buffer
+ stream_.convertInfo[mode].inJump = stream_.nUserChannels[0];
+ stream_.convertInfo[mode].outJump = stream_.nDeviceChannels[0];
+ stream_.convertInfo[mode].inFormat = stream_.userFormat;
+ stream_.convertInfo[mode].outFormat = stream_.deviceFormat[0];
}
- void RtApi :: setConvertInfo( StreamMode mode, unsigned int firstChannel )
- {
- if ( mode == INPUT ) { // convert device to user buffer
- stream_.convertInfo[mode].inJump = stream_.nDeviceChannels[1];
- stream_.convertInfo[mode].outJump = stream_.nUserChannels[1];
- stream_.convertInfo[mode].inFormat = stream_.deviceFormat[1];
- stream_.convertInfo[mode].outFormat = stream_.userFormat;
+ if ( stream_.convertInfo[mode].inJump < stream_.convertInfo[mode].outJump )
+ stream_.convertInfo[mode].channels = stream_.convertInfo[mode].inJump;
+ else
+ stream_.convertInfo[mode].channels = stream_.convertInfo[mode].outJump;
+
+ // Set up the interleave/deinterleave offsets.
+ if ( stream_.deviceInterleaved[mode] != stream_.userInterleaved ) {
+ if ( ( mode == OUTPUT && stream_.deviceInterleaved[mode] ) ||
+ ( mode == INPUT && stream_.userInterleaved ) ) {
+ for ( int k=0; k<stream_.convertInfo[mode].channels; k++ ) {
+ stream_.convertInfo[mode].inOffset.push_back( k * stream_.bufferSize );
+ stream_.convertInfo[mode].outOffset.push_back( k );
+ stream_.convertInfo[mode].inJump = 1;
+ }
}
- else { // convert user to device buffer
- stream_.convertInfo[mode].inJump = stream_.nUserChannels[0];
- stream_.convertInfo[mode].outJump = stream_.nDeviceChannels[0];
- stream_.convertInfo[mode].inFormat = stream_.userFormat;
- stream_.convertInfo[mode].outFormat = stream_.deviceFormat[0];
+ else {
+ for ( int k=0; k<stream_.convertInfo[mode].channels; k++ ) {
+ stream_.convertInfo[mode].inOffset.push_back( k );
+ stream_.convertInfo[mode].outOffset.push_back( k * stream_.bufferSize );
+ stream_.convertInfo[mode].outJump = 1;
+ }
}
-
- if ( stream_.convertInfo[mode].inJump < stream_.convertInfo[mode].outJump )
- stream_.convertInfo[mode].channels = stream_.convertInfo[mode].inJump;
- else
- stream_.convertInfo[mode].channels = stream_.convertInfo[mode].outJump;
-
- // Set up the interleave/deinterleave offsets.
- if ( stream_.deviceInterleaved[mode] != stream_.userInterleaved ) {
- if ( ( mode == OUTPUT && stream_.deviceInterleaved[mode] ) ||
- ( mode == INPUT && stream_.userInterleaved ) ) {
- for ( int k=0; k<stream_.convertInfo[mode].channels; k++ ) {
- stream_.convertInfo[mode].inOffset.push_back( k * stream_.bufferSize );
- stream_.convertInfo[mode].outOffset.push_back( k );
- stream_.convertInfo[mode].inJump = 1;
- }
+ }
+ else { // no (de)interleaving
+ if ( stream_.userInterleaved ) {
+ for ( int k=0; k<stream_.convertInfo[mode].channels; k++ ) {
+ stream_.convertInfo[mode].inOffset.push_back( k );
+ stream_.convertInfo[mode].outOffset.push_back( k );
}
- else {
- for ( int k=0; k<stream_.convertInfo[mode].channels; k++ ) {
- stream_.convertInfo[mode].inOffset.push_back( k );
- stream_.convertInfo[mode].outOffset.push_back( k * stream_.bufferSize );
- stream_.convertInfo[mode].outJump = 1;
- }
+ }
+ else {
+ for ( int k=0; k<stream_.convertInfo[mode].channels; k++ ) {
+ stream_.convertInfo[mode].inOffset.push_back( k * stream_.bufferSize );
+ stream_.convertInfo[mode].outOffset.push_back( k * stream_.bufferSize );
+ stream_.convertInfo[mode].inJump = 1;
+ stream_.convertInfo[mode].outJump = 1;
}
}
- else { // no (de)interleaving
- if ( stream_.userInterleaved ) {
- for ( int k=0; k<stream_.convertInfo[mode].channels; k++ ) {
- stream_.convertInfo[mode].inOffset.push_back( k );
- stream_.convertInfo[mode].outOffset.push_back( k );
- }
+ }
+
+ // Add channel offset.
+ if ( firstChannel > 0 ) {
+ if ( stream_.deviceInterleaved[mode] ) {
+ if ( mode == OUTPUT ) {
+ for ( int k=0; k<stream_.convertInfo[mode].channels; k++ )
+ stream_.convertInfo[mode].outOffset[k] += firstChannel;
}
else {
- for ( int k=0; k<stream_.convertInfo[mode].channels; k++ ) {
- stream_.convertInfo[mode].inOffset.push_back( k * stream_.bufferSize );
- stream_.convertInfo[mode].outOffset.push_back( k * stream_.bufferSize );
- stream_.convertInfo[mode].inJump = 1;
- stream_.convertInfo[mode].outJump = 1;
- }
+ for ( int k=0; k<stream_.convertInfo[mode].channels; k++ )
+ stream_.convertInfo[mode].inOffset[k] += firstChannel;
}
}
-
- // Add channel offset.
- if ( firstChannel > 0 ) {
- if ( stream_.deviceInterleaved[mode] ) {
- if ( mode == OUTPUT ) {
- for ( int k=0; k<stream_.convertInfo[mode].channels; k++ )
- stream_.convertInfo[mode].outOffset[k] += firstChannel;
- }
- else {
- for ( int k=0; k<stream_.convertInfo[mode].channels; k++ )
- stream_.convertInfo[mode].inOffset[k] += firstChannel;
- }
+ else {
+ if ( mode == OUTPUT ) {
+ for ( int k=0; k<stream_.convertInfo[mode].channels; k++ )
+ stream_.convertInfo[mode].outOffset[k] += ( firstChannel * stream_.bufferSize );
}
else {
- if ( mode == OUTPUT ) {
- for ( int k=0; k<stream_.convertInfo[mode].channels; k++ )
- stream_.convertInfo[mode].outOffset[k] += ( firstChannel * stream_.bufferSize );
- }
- else {
- for ( int k=0; k<stream_.convertInfo[mode].channels; k++ )
- stream_.convertInfo[mode].inOffset[k] += ( firstChannel * stream_.bufferSize );
- }
+ for ( int k=0; k<stream_.convertInfo[mode].channels; k++ )
+ stream_.convertInfo[mode].inOffset[k] += ( firstChannel * stream_.bufferSize );
}
}
}
+}
- void RtApi :: convertBuffer( char *outBuffer, char *inBuffer, ConvertInfo &info )
- {
- // This function does format conversion, input/output channel compensation, and
- // data interleaving/deinterleaving. 24-bit integers are assumed to occupy
- // the upper three bytes of a 32-bit integer.
-
- // Clear our device buffer when in/out duplex device channels are different
- if ( outBuffer == stream_.deviceBuffer && stream_.mode == DUPLEX &&
- ( stream_.nDeviceChannels[0] < stream_.nDeviceChannels[1] ) )
- memset( outBuffer, 0, stream_.bufferSize * info.outJump * formatBytes( info.outFormat ) );
-
- int j;
- if (info.outFormat == RTAUDIO_FLOAT64) {
- Float64 scale;
- Float64 *out = (Float64 *)outBuffer;
-
- if (info.inFormat == RTAUDIO_SINT8) {
- signed char *in = (signed char *)inBuffer;
- scale = 1.0 / 127.5;
- for (unsigned int i=0; i<stream_.bufferSize; i++) {
- for (j=0; j<info.channels; j++) {
- out[info.outOffset[j]] = (Float64) in[info.inOffset[j]];
- out[info.outOffset[j]] += 0.5;
- out[info.outOffset[j]] *= scale;
- }
- in += info.inJump;
- out += info.outJump;
+void RtApi :: convertBuffer( char *outBuffer, char *inBuffer, ConvertInfo &info )
+{
+ // This function does format conversion, input/output channel compensation, and
+ // data interleaving/deinterleaving. 24-bit integers are assumed to occupy
+ // the upper three bytes of a 32-bit integer.
+
+ // Clear our device buffer when in/out duplex device channels are different
+ if ( outBuffer == stream_.deviceBuffer && stream_.mode == DUPLEX &&
+ ( stream_.nDeviceChannels[0] < stream_.nDeviceChannels[1] ) )
+ memset( outBuffer, 0, stream_.bufferSize * info.outJump * formatBytes( info.outFormat ) );
+
+ int j;
+ if (info.outFormat == RTAUDIO_FLOAT64) {
+ Float64 scale;
+ Float64 *out = (Float64 *)outBuffer;
+
+ if (info.inFormat == RTAUDIO_SINT8) {
+ signed char *in = (signed char *)inBuffer;
+ scale = 1.0 / 127.5;
+ for (unsigned int i=0; i<stream_.bufferSize; i++) {
+ for (j=0; j<info.channels; j++) {
+ out[info.outOffset[j]] = (Float64) in[info.inOffset[j]];
+ out[info.outOffset[j]] += 0.5;
+ out[info.outOffset[j]] *= scale;
}
- }
- else if (info.inFormat == RTAUDIO_SINT16) {
- Int16 *in = (Int16 *)inBuffer;
- scale = 1.0 / 32767.5;
- for (unsigned int i=0; i<stream_.bufferSize; i++) {
- for (j=0; j<info.channels; j++) {
- out[info.outOffset[j]] = (Float64) in[info.inOffset[j]];
- out[info.outOffset[j]] += 0.5;
- out[info.outOffset[j]] *= scale;
- }
- in += info.inJump;
- out += info.outJump;
+ in += info.inJump;
+ out += info.outJump;
+ }
+ }
+ else if (info.inFormat == RTAUDIO_SINT16) {
+ Int16 *in = (Int16 *)inBuffer;
+ scale = 1.0 / 32767.5;
+ for (unsigned int i=0; i<stream_.bufferSize; i++) {
+ for (j=0; j<info.channels; j++) {
+ out[info.outOffset[j]] = (Float64) in[info.inOffset[j]];
+ out[info.outOffset[j]] += 0.5;
+ out[info.outOffset[j]] *= scale;
}
- }
- else if (info.inFormat == RTAUDIO_SINT24) {
- Int32 *in = (Int32 *)inBuffer;
- scale = 1.0 / 8388607.5;
- for (unsigned int i=0; i<stream_.bufferSize; i++) {
- for (j=0; j<info.channels; j++) {
- out[info.outOffset[j]] = (Float64) (in[info.inOffset[j]] & 0x00ffffff);
- out[info.outOffset[j]] += 0.5;
- out[info.outOffset[j]] *= scale;
- }
- in += info.inJump;
- out += info.outJump;
+ in += info.inJump;
+ out += info.outJump;
+ }
+ }
+ else if (info.inFormat == RTAUDIO_SINT24) {
+ Int32 *in = (Int32 *)inBuffer;
+ scale = 1.0 / 8388607.5;
+ for (unsigned int i=0; i<stream_.bufferSize; i++) {
+ for (j=0; j<info.channels; j++) {
+ out[info.outOffset[j]] = (Float64) (in[info.inOffset[j]] & 0x00ffffff);
+ out[info.outOffset[j]] += 0.5;
+ out[info.outOffset[j]] *= scale;
}
- }
- else if (info.inFormat == RTAUDIO_SINT32) {
- Int32 *in = (Int32 *)inBuffer;
- scale = 1.0 / 2147483647.5;
- for (unsigned int i=0; i<stream_.bufferSize; i++) {
- for (j=0; j<info.channels; j++) {
- out[info.outOffset[j]] = (Float64) in[info.inOffset[j]];
- out[info.outOffset[j]] += 0.5;
- out[info.outOffset[j]] *= scale;
- }
- in += info.inJump;
- out += info.outJump;
+ in += info.inJump;
+ out += info.outJump;
+ }
+ }
+ else if (info.inFormat == RTAUDIO_SINT32) {
+ Int32 *in = (Int32 *)inBuffer;
+ scale = 1.0 / 2147483647.5;
+ for (unsigned int i=0; i<stream_.bufferSize; i++) {
+ for (j=0; j<info.channels; j++) {
+ out[info.outOffset[j]] = (Float64) in[info.inOffset[j]];
+ out[info.outOffset[j]] += 0.5;
+ out[info.outOffset[j]] *= scale;
}
+ in += info.inJump;
+ out += info.outJump;
}
- else if (info.inFormat == RTAUDIO_FLOAT32) {
- Float32 *in = (Float32 *)inBuffer;
- for (unsigned int i=0; i<stream_.bufferSize; i++) {
- for (j=0; j<info.channels; j++) {
- out[info.outOffset[j]] = (Float64) in[info.inOffset[j]];
- }
- in += info.inJump;
- out += info.outJump;
+ }
+ else if (info.inFormat == RTAUDIO_FLOAT32) {
+ Float32 *in = (Float32 *)inBuffer;
+ for (unsigned int i=0; i<stream_.bufferSize; i++) {
+ for (j=0; j<info.channels; j++) {
+ out[info.outOffset[j]] = (Float64) in[info.inOffset[j]];
}
+ in += info.inJump;
+ out += info.outJump;
}
- else if (info.inFormat == RTAUDIO_FLOAT64) {
- // Channel compensation and/or (de)interleaving only.
- Float64 *in = (Float64 *)inBuffer;
- for (unsigned int i=0; i<stream_.bufferSize; i++) {
- for (j=0; j<info.channels; j++) {
- out[info.outOffset[j]] = in[info.inOffset[j]];
- }
- in += info.inJump;
- out += info.outJump;
+ }
+ else if (info.inFormat == RTAUDIO_FLOAT64) {
+ // Channel compensation and/or (de)interleaving only.
+ Float64 *in = (Float64 *)inBuffer;
+ for (unsigned int i=0; i<stream_.bufferSize; i++) {
+ for (j=0; j<info.channels; j++) {
+ out[info.outOffset[j]] = in[info.inOffset[j]];
}
+ in += info.inJump;
+ out += info.outJump;
}
}
- else if (info.outFormat == RTAUDIO_FLOAT32) {
- Float32 scale;
- Float32 *out = (Float32 *)outBuffer;
-
- if (info.inFormat == RTAUDIO_SINT8) {
- signed char *in = (signed char *)inBuffer;
- scale = (Float32) ( 1.0 / 127.5 );
- for (unsigned int i=0; i<stream_.bufferSize; i++) {
- for (j=0; j<info.channels; j++) {
- out[info.outOffset[j]] = (Float32) in[info.inOffset[j]];
- out[info.outOffset[j]] += 0.5;
- out[info.outOffset[j]] *= scale;
- }
- in += info.inJump;
- out += info.outJump;
+ }
+ else if (info.outFormat == RTAUDIO_FLOAT32) {
+ Float32 scale;
+ Float32 *out = (Float32 *)outBuffer;
+
+ if (info.inFormat == RTAUDIO_SINT8) {
+ signed char *in = (signed char *)inBuffer;
+ scale = (Float32) ( 1.0 / 127.5 );
+ for (unsigned int i=0; i<stream_.bufferSize; i++) {
+ for (j=0; j<info.channels; j++) {
+ out[info.outOffset[j]] = (Float32) in[info.inOffset[j]];
+ out[info.outOffset[j]] += 0.5;
+ out[info.outOffset[j]] *= scale;
}
- }
- else if (info.inFormat == RTAUDIO_SINT16) {
- Int16 *in = (Int16 *)inBuffer;
- scale = (Float32) ( 1.0 / 32767.5 );
- for (unsigned int i=0; i<stream_.bufferSize; i++) {
- for (j=0; j<info.channels; j++) {
- out[info.outOffset[j]] = (Float32) in[info.inOffset[j]];
- out[info.outOffset[j]] += 0.5;
- out[info.outOffset[j]] *= scale;
- }
- in += info.inJump;
- out += info.outJump;
+ in += info.inJump;
+ out += info.outJump;
+ }
+ }
+ else if (info.inFormat == RTAUDIO_SINT16) {
+ Int16 *in = (Int16 *)inBuffer;
+ scale = (Float32) ( 1.0 / 32767.5 );
+ for (unsigned int i=0; i<stream_.bufferSize; i++) {
+ for (j=0; j<info.channels; j++) {
+ out[info.outOffset[j]] = (Float32) in[info.inOffset[j]];
+ out[info.outOffset[j]] += 0.5;
+ out[info.outOffset[j]] *= scale;
}
- }
- else if (info.inFormat == RTAUDIO_SINT24) {
- Int32 *in = (Int32 *)inBuffer;
- scale = (Float32) ( 1.0 / 8388607.5 );
- for (unsigned int i=0; i<stream_.bufferSize; i++) {
- for (j=0; j<info.channels; j++) {
- out[info.outOffset[j]] = (Float32) (in[info.inOffset[j]] & 0x00ffffff);
- out[info.outOffset[j]] += 0.5;
- out[info.outOffset[j]] *= scale;
- }
- in += info.inJump;
- out += info.outJump;
+ in += info.inJump;
+ out += info.outJump;
+ }
+ }
+ else if (info.inFormat == RTAUDIO_SINT24) {
+ Int32 *in = (Int32 *)inBuffer;
+ scale = (Float32) ( 1.0 / 8388607.5 );
+ for (unsigned int i=0; i<stream_.bufferSize; i++) {
+ for (j=0; j<info.channels; j++) {
+ out[info.outOffset[j]] = (Float32) (in[info.inOffset[j]] & 0x00ffffff);
+ out[info.outOffset[j]] += 0.5;
+ out[info.outOffset[j]] *= scale;
}
- }
- else if (info.inFormat == RTAUDIO_SINT32) {
- Int32 *in = (Int32 *)inBuffer;
- scale = (Float32) ( 1.0 / 2147483647.5 );
- for (unsigned int i=0; i<stream_.bufferSize; i++) {
- for (j=0; j<info.channels; j++) {
- out[info.outOffset[j]] = (Float32) in[info.inOffset[j]];
- out[info.outOffset[j]] += 0.5;
- out[info.outOffset[j]] *= scale;
- }
- in += info.inJump;
- out += info.outJump;
+ in += info.inJump;
+ out += info.outJump;
+ }
+ }
+ else if (info.inFormat == RTAUDIO_SINT32) {
+ Int32 *in = (Int32 *)inBuffer;
+ scale = (Float32) ( 1.0 / 2147483647.5 );
+ for (unsigned int i=0; i<stream_.bufferSize; i++) {
+ for (j=0; j<info.channels; j++) {
+ out[info.outOffset[j]] = (Float32) in[info.inOffset[j]];
+ out[info.outOffset[j]] += 0.5;
+ out[info.outOffset[j]] *= scale;
}
+ in += info.inJump;
+ out += info.outJump;
}
- else if (info.inFormat == RTAUDIO_FLOAT32) {
- // Channel compensation and/or (de)interleaving only.
- Float32 *in = (Float32 *)inBuffer;
- for (unsigned int i=0; i<stream_.bufferSize; i++) {
- for (j=0; j<info.channels; j++) {
- out[info.outOffset[j]] = in[info.inOffset[j]];
- }
- in += info.inJump;
- out += info.outJump;
+ }
+ else if (info.inFormat == RTAUDIO_FLOAT32) {
+ // Channel compensation and/or (de)interleaving only.
+ Float32 *in = (Float32 *)inBuffer;
+ for (unsigned int i=0; i<stream_.bufferSize; i++) {
+ for (j=0; j<info.channels; j++) {
+ out[info.outOffset[j]] = in[info.inOffset[j]];
}
+ in += info.inJump;
+ out += info.outJump;
}
- else if (info.inFormat == RTAUDIO_FLOAT64) {
- Float64 *in = (Float64 *)inBuffer;
- for (unsigned int i=0; i<stream_.bufferSize; i++) {
- for (j=0; j<info.channels; j++) {
- out[info.outOffset[j]] = (Float32) in[info.inOffset[j]];
- }
- in += info.inJump;
- out += info.outJump;
+ }
+ else if (info.inFormat == RTAUDIO_FLOAT64) {
+ Float64 *in = (Float64 *)inBuffer;
+ for (unsigned int i=0; i<stream_.bufferSize; i++) {
+ for (j=0; j<info.channels; j++) {
+ out[info.outOffset[j]] = (Float32) in[info.inOffset[j]];
}
+ in += info.inJump;
+ out += info.outJump;
}
}
- else if (info.outFormat == RTAUDIO_SINT32) {
- Int32 *out = (Int32 *)outBuffer;
- if (info.inFormat == RTAUDIO_SINT8) {
- signed char *in = (signed char *)inBuffer;
- for (unsigned int i=0; i<stream_.bufferSize; i++) {
- for (j=0; j<info.channels; j++) {
- out[info.outOffset[j]] = (Int32) in[info.inOffset[j]];
- out[info.outOffset[j]] <<= 24;
- }
- in += info.inJump;
- out += info.outJump;
+ }
+ else if (info.outFormat == RTAUDIO_SINT32) {
+ Int32 *out = (Int32 *)outBuffer;
+ if (info.inFormat == RTAUDIO_SINT8) {
+ signed char *in = (signed char *)inBuffer;
+ for (unsigned int i=0; i<stream_.bufferSize; i++) {
+ for (j=0; j<info.channels; j++) {
+ out[info.outOffset[j]] = (Int32) in[info.inOffset[j]];
+ out[info.outOffset[j]] <<= 24;
}
+ in += info.inJump;
+ out += info.outJump;
}
- else if (info.inFormat == RTAUDIO_SINT16) {
- Int16 *in = (Int16 *)inBuffer;
- for (unsigned int i=0; i<stream_.bufferSize; i++) {
- for (j=0; j<info.channels; j++) {
- out[info.outOffset[j]] = (Int32) in[info.inOffset[j]];
- out[info.outOffset[j]] <<= 16;
- }
- in += info.inJump;
- out += info.outJump;
+ }
+ else if (info.inFormat == RTAUDIO_SINT16) {
+ Int16 *in = (Int16 *)inBuffer;
+ for (unsigned int i=0; i<stream_.bufferSize; i++) {
+ for (j=0; j<info.channels; j++) {
+ out[info.outOffset[j]] = (Int32) in[info.inOffset[j]];
+ out[info.outOffset[j]] <<= 16;
}
+ in += info.inJump;
+ out += info.outJump;
}
- else if (info.inFormat == RTAUDIO_SINT24) {
- Int32 *in = (Int32 *)inBuffer;
- for (unsigned int i=0; i<stream_.bufferSize; i++) {
- for (j=0; j<info.channels; j++) {
- out[info.outOffset[j]] = (Int32) in[info.inOffset[j]];
- out[info.outOffset[j]] <<= 8;
- }
- in += info.inJump;
- out += info.outJump;
+ }
+ else if (info.inFormat == RTAUDIO_SINT24) {
+ Int32 *in = (Int32 *)inBuffer;
+ for (unsigned int i=0; i<stream_.bufferSize; i++) {
+ for (j=0; j<info.channels; j++) {
+ out[info.outOffset[j]] = (Int32) in[info.inOffset[j]];
+ out[info.outOffset[j]] <<= 8;
}
+ in += info.inJump;
+ out += info.outJump;
}
- else if (info.inFormat == RTAUDIO_SINT32) {
- // Channel compensation and/or (de)interleaving only.
- Int32 *in = (Int32 *)inBuffer;
- for (unsigned int i=0; i<stream_.bufferSize; i++) {
- for (j=0; j<info.channels; j++) {
- out[info.outOffset[j]] = in[info.inOffset[j]];
- }
- in += info.inJump;
- out += info.outJump;
+ }
+ else if (info.inFormat == RTAUDIO_SINT32) {
+ // Channel compensation and/or (de)interleaving only.
+ Int32 *in = (Int32 *)inBuffer;
+ for (unsigned int i=0; i<stream_.bufferSize; i++) {
+ for (j=0; j<info.channels; j++) {
+ out[info.outOffset[j]] = in[info.inOffset[j]];
}
+ in += info.inJump;
+ out += info.outJump;
}
- else if (info.inFormat == RTAUDIO_FLOAT32) {
- Float32 *in = (Float32 *)inBuffer;
- for (unsigned int i=0; i<stream_.bufferSize; i++) {
- for (j=0; j<info.channels; j++) {
- out[info.outOffset[j]] = (Int32) (in[info.inOffset[j]] * 2147483647.5 - 0.5);
- }
- in += info.inJump;
- out += info.outJump;
+ }
+ else if (info.inFormat == RTAUDIO_FLOAT32) {
+ Float32 *in = (Float32 *)inBuffer;
+ for (unsigned int i=0; i<stream_.bufferSize; i++) {
+ for (j=0; j<info.channels; j++) {
+ out[info.outOffset[j]] = (Int32) (in[info.inOffset[j]] * 2147483647.5 - 0.5);
}
+ in += info.inJump;
+ out += info.outJump;
}
- else if (info.inFormat == RTAUDIO_FLOAT64) {
- Float64 *in = (Float64 *)inBuffer;
- for (unsigned int i=0; i<stream_.bufferSize; i++) {
- for (j=0; j<info.channels; j++) {
- out[info.outOffset[j]] = (Int32) (in[info.inOffset[j]] * 2147483647.5 - 0.5);
- }
- in += info.inJump;
- out += info.outJump;
+ }
+ else if (info.inFormat == RTAUDIO_FLOAT64) {
+ Float64 *in = (Float64 *)inBuffer;
+ for (unsigned int i=0; i<stream_.bufferSize; i++) {
+ for (j=0; j<info.channels; j++) {
+ out[info.outOffset[j]] = (Int32) (in[info.inOffset[j]] * 2147483647.5 - 0.5);
}
+ in += info.inJump;
+ out += info.outJump;
}
}
- else if (info.outFormat == RTAUDIO_SINT24) {
- Int32 *out = (Int32 *)outBuffer;
- if (info.inFormat == RTAUDIO_SINT8) {
- signed char *in = (signed char *)inBuffer;
- for (unsigned int i=0; i<stream_.bufferSize; i++) {
- for (j=0; j<info.channels; j++) {
- out[info.outOffset[j]] = (Int32) in[info.inOffset[j]];
- out[info.outOffset[j]] <<= 16;
- }
- in += info.inJump;
- out += info.outJump;
+ }
+ else if (info.outFormat == RTAUDIO_SINT24) {
+ Int32 *out = (Int32 *)outBuffer;
+ if (info.inFormat == RTAUDIO_SINT8) {
+ signed char *in = (signed char *)inBuffer;
+ for (unsigned int i=0; i<stream_.bufferSize; i++) {
+ for (j=0; j<info.channels; j++) {
+ out[info.outOffset[j]] = (Int32) in[info.inOffset[j]];
+ out[info.outOffset[j]] <<= 16;
}
+ in += info.inJump;
+ out += info.outJump;
}
- else if (info.inFormat == RTAUDIO_SINT16) {
- Int16 *in = (Int16 *)inBuffer;
- for (unsigned int i=0; i<stream_.bufferSize; i++) {
- for (j=0; j<info.channels; j++) {
- out[info.outOffset[j]] = (Int32) in[info.inOffset[j]];
- out[info.outOffset[j]] <<= 8;
- }
- in += info.inJump;
- out += info.outJump;
+ }
+ else if (info.inFormat == RTAUDIO_SINT16) {
+ Int16 *in = (Int16 *)inBuffer;
+ for (unsigned int i=0; i<stream_.bufferSize; i++) {
+ for (j=0; j<info.channels; j++) {
+ out[info.outOffset[j]] = (Int32) in[info.inOffset[j]];
+ out[info.outOffset[j]] <<= 8;
}
+ in += info.inJump;
+ out += info.outJump;
}
- else if (info.inFormat == RTAUDIO_SINT24) {
- // Channel compensation and/or (de)interleaving only.
- Int32 *in = (Int32 *)inBuffer;
- for (unsigned int i=0; i<stream_.bufferSize; i++) {
- for (j=0; j<info.channels; j++) {
- out[info.outOffset[j]] = in[info.inOffset[j]];
- }
- in += info.inJump;
- out += info.outJump;
+ }
+ else if (info.inFormat == RTAUDIO_SINT24) {
+ // Channel compensation and/or (de)interleaving only.
+ Int32 *in = (Int32 *)inBuffer;
+ for (unsigned int i=0; i<stream_.bufferSize; i++) {
+ for (j=0; j<info.channels; j++) {
+ out[info.outOffset[j]] = in[info.inOffset[j]];
}
+ in += info.inJump;
+ out += info.outJump;
}
- else if (info.inFormat == RTAUDIO_SINT32) {
- Int32 *in = (Int32 *)inBuffer;
- for (unsigned int i=0; i<stream_.bufferSize; i++) {
- for (j=0; j<info.channels; j++) {
- out[info.outOffset[j]] = (Int32) in[info.inOffset[j]];
- out[info.outOffset[j]] >>= 8;
- }
- in += info.inJump;
- out += info.outJump;
+ }
+ else if (info.inFormat == RTAUDIO_SINT32) {
+ Int32 *in = (Int32 *)inBuffer;
+ for (unsigned int i=0; i<stream_.bufferSize; i++) {
+ for (j=0; j<info.channels; j++) {
+ out[info.outOffset[j]] = (Int32) in[info.inOffset[j]];
+ out[info.outOffset[j]] >>= 8;
}
+ in += info.inJump;
+ out += info.outJump;
}
- else if (info.inFormat == RTAUDIO_FLOAT32) {
- Float32 *in = (Float32 *)inBuffer;
- for (unsigned int i=0; i<stream_.bufferSize; i++) {
- for (j=0; j<info.channels; j++) {
- out[info.outOffset[j]] = (Int32) (in[info.inOffset[j]] * 8388607.5 - 0.5);
- }
- in += info.inJump;
- out += info.outJump;
+ }
+ else if (info.inFormat == RTAUDIO_FLOAT32) {
+ Float32 *in = (Float32 *)inBuffer;
+ for (unsigned int i=0; i<stream_.bufferSize; i++) {
+ for (j=0; j<info.channels; j++) {
+ out[info.outOffset[j]] = (Int32) (in[info.inOffset[j]] * 8388607.5 - 0.5);
}
+ in += info.inJump;
+ out += info.outJump;
}
- else if (info.inFormat == RTAUDIO_FLOAT64) {
- Float64 *in = (Float64 *)inBuffer;
- for (unsigned int i=0; i<stream_.bufferSize; i++) {
- for (j=0; j<info.channels; j++) {
- out[info.outOffset[j]] = (Int32) (in[info.inOffset[j]] * 8388607.5 - 0.5);
- }
- in += info.inJump;
- out += info.outJump;
+ }
+ else if (info.inFormat == RTAUDIO_FLOAT64) {
+ Float64 *in = (Float64 *)inBuffer;
+ for (unsigned int i=0; i<stream_.bufferSize; i++) {
+ for (j=0; j<info.channels; j++) {
+ out[info.outOffset[j]] = (Int32) (in[info.inOffset[j]] * 8388607.5 - 0.5);
}
+ in += info.inJump;
+ out += info.outJump;
}
}
- else if (info.outFormat == RTAUDIO_SINT16) {
- Int16 *out = (Int16 *)outBuffer;
- if (info.inFormat == RTAUDIO_SINT8) {
- signed char *in = (signed char *)inBuffer;
- for (unsigned int i=0; i<stream_.bufferSize; i++) {
- for (j=0; j<info.channels; j++) {
- out[info.outOffset[j]] = (Int16) in[info.inOffset[j]];
- out[info.outOffset[j]] <<= 8;
- }
- in += info.inJump;
- out += info.outJump;
+ }
+ else if (info.outFormat == RTAUDIO_SINT16) {
+ Int16 *out = (Int16 *)outBuffer;
+ if (info.inFormat == RTAUDIO_SINT8) {
+ signed char *in = (signed char *)inBuffer;
+ for (unsigned int i=0; i<stream_.bufferSize; i++) {
+ for (j=0; j<info.channels; j++) {
+ out[info.outOffset[j]] = (Int16) in[info.inOffset[j]];
+ out[info.outOffset[j]] <<= 8;
}
+ in += info.inJump;
+ out += info.outJump;
}
- else if (info.inFormat == RTAUDIO_SINT16) {
- // Channel compensation and/or (de)interleaving only.
- Int16 *in = (Int16 *)inBuffer;
- for (unsigned int i=0; i<stream_.bufferSize; i++) {
- for (j=0; j<info.channels; j++) {
- out[info.outOffset[j]] = in[info.inOffset[j]];
- }
- in += info.inJump;
- out += info.outJump;
+ }
+ else if (info.inFormat == RTAUDIO_SINT16) {
+ // Channel compensation and/or (de)interleaving only.
+ Int16 *in = (Int16 *)inBuffer;
+ for (unsigned int i=0; i<stream_.bufferSize; i++) {
+ for (j=0; j<info.channels; j++) {
+ out[info.outOffset[j]] = in[info.inOffset[j]];
}
+ in += info.inJump;
+ out += info.outJump;
}
- else if (info.inFormat == RTAUDIO_SINT24) {
- Int32 *in = (Int32 *)inBuffer;
- for (unsigned int i=0; i<stream_.bufferSize; i++) {
- for (j=0; j<info.channels; j++) {
- out[info.outOffset[j]] = (Int16) ((in[info.inOffset[j]] >> 8) & 0x0000ffff);
- }
- in += info.inJump;
- out += info.outJump;
+ }
+ else if (info.inFormat == RTAUDIO_SINT24) {
+ Int32 *in = (Int32 *)inBuffer;
+ for (unsigned int i=0; i<stream_.bufferSize; i++) {
+ for (j=0; j<info.channels; j++) {
+ out[info.outOffset[j]] = (Int16) ((in[info.inOffset[j]] >> 8) & 0x0000ffff);
}
+ in += info.inJump;
+ out += info.outJump;
}
- else if (info.inFormat == RTAUDIO_SINT32) {
- Int32 *in = (Int32 *)inBuffer;
- for (unsigned int i=0; i<stream_.bufferSize; i++) {
- for (j=0; j<info.channels; j++) {
- out[info.outOffset[j]] = (Int16) ((in[info.inOffset[j]] >> 16) & 0x0000ffff);
- }
- in += info.inJump;
- out += info.outJump;
+ }
+ else if (info.inFormat == RTAUDIO_SINT32) {
+ Int32 *in = (Int32 *)inBuffer;
+ for (unsigned int i=0; i<stream_.bufferSize; i++) {
+ for (j=0; j<info.channels; j++) {
+ out[info.outOffset[j]] = (Int16) ((in[info.inOffset[j]] >> 16) & 0x0000ffff);
}
+ in += info.inJump;
+ out += info.outJump;
}
- else if (info.inFormat == RTAUDIO_FLOAT32) {
- Float32 *in = (Float32 *)inBuffer;
- for (unsigned int i=0; i<stream_.bufferSize; i++) {
- for (j=0; j<info.channels; j++) {
- out[info.outOffset[j]] = (Int16) (in[info.inOffset[j]] * 32767.5 - 0.5);
- }
- in += info.inJump;
- out += info.outJump;
+ }
+ else if (info.inFormat == RTAUDIO_FLOAT32) {
+ Float32 *in = (Float32 *)inBuffer;
+ for (unsigned int i=0; i<stream_.bufferSize; i++) {
+ for (j=0; j<info.channels; j++) {
+ out[info.outOffset[j]] = (Int16) (in[info.inOffset[j]] * 32767.5 - 0.5);
}
+ in += info.inJump;
+ out += info.outJump;
}
- else if (info.inFormat == RTAUDIO_FLOAT64) {
- Float64 *in = (Float64 *)inBuffer;
- for (unsigned int i=0; i<stream_.bufferSize; i++) {
- for (j=0; j<info.channels; j++) {
- out[info.outOffset[j]] = (Int16) (in[info.inOffset[j]] * 32767.5 - 0.5);
- }
- in += info.inJump;
- out += info.outJump;
+ }
+ else if (info.inFormat == RTAUDIO_FLOAT64) {
+ Float64 *in = (Float64 *)inBuffer;
+ for (unsigned int i=0; i<stream_.bufferSize; i++) {
+ for (j=0; j<info.channels; j++) {
+ out[info.outOffset[j]] = (Int16) (in[info.inOffset[j]] * 32767.5 - 0.5);
}
+ in += info.inJump;
+ out += info.outJump;
}
}
- else if (info.outFormat == RTAUDIO_SINT8) {
- signed char *out = (signed char *)outBuffer;
- if (info.inFormat == RTAUDIO_SINT8) {
- // Channel compensation and/or (de)interleaving only.
- signed char *in = (signed char *)inBuffer;
- for (unsigned int i=0; i<stream_.bufferSize; i++) {
- for (j=0; j<info.channels; j++) {
- out[info.outOffset[j]] = in[info.inOffset[j]];
- }
- in += info.inJump;
- out += info.outJump;
+ }
+ else if (info.outFormat == RTAUDIO_SINT8) {
+ signed char *out = (signed char *)outBuffer;
+ if (info.inFormat == RTAUDIO_SINT8) {
+ // Channel compensation and/or (de)interleaving only.
+ signed char *in = (signed char *)inBuffer;
+ for (unsigned int i=0; i<stream_.bufferSize; i++) {
+ for (j=0; j<info.channels; j++) {
+ out[info.outOffset[j]] = in[info.inOffset[j]];
}
+ in += info.inJump;
+ out += info.outJump;
}
- if (info.inFormat == RTAUDIO_SINT16) {
- Int16 *in = (Int16 *)inBuffer;
- for (unsigned int i=0; i<stream_.bufferSize; i++) {
- for (j=0; j<info.channels; j++) {
- out[info.outOffset[j]] = (signed char) ((in[info.inOffset[j]] >> 8) & 0x00ff);
- }
- in += info.inJump;
- out += info.outJump;
+ }
+ if (info.inFormat == RTAUDIO_SINT16) {
+ Int16 *in = (Int16 *)inBuffer;
+ for (unsigned int i=0; i<stream_.bufferSize; i++) {
+ for (j=0; j<info.channels; j++) {
+ out[info.outOffset[j]] = (signed char) ((in[info.inOffset[j]] >> 8) & 0x00ff);
}
+ in += info.inJump;
+ out += info.outJump;
}
- else if (info.inFormat == RTAUDIO_SINT24) {
- Int32 *in = (Int32 *)inBuffer;
- for (unsigned int i=0; i<stream_.bufferSize; i++) {
- for (j=0; j<info.channels; j++) {
- out[info.outOffset[j]] = (signed char) ((in[info.inOffset[j]] >> 16) & 0x000000ff);
- }
- in += info.inJump;
- out += info.outJump;
+ }
+ else if (info.inFormat == RTAUDIO_SINT24) {
+ Int32 *in = (Int32 *)inBuffer;
+ for (unsigned int i=0; i<stream_.bufferSize; i++) {
+ for (j=0; j<info.channels; j++) {
+ out[info.outOffset[j]] = (signed char) ((in[info.inOffset[j]] >> 16) & 0x000000ff);
}
+ in += info.inJump;
+ out += info.outJump;
}
- else if (info.inFormat == RTAUDIO_SINT32) {
- Int32 *in = (Int32 *)inBuffer;
- for (unsigned int i=0; i<stream_.bufferSize; i++) {
- for (j=0; j<info.channels; j++) {
- out[info.outOffset[j]] = (signed char) ((in[info.inOffset[j]] >> 24) & 0x000000ff);
- }
- in += info.inJump;
- out += info.outJump;
+ }
+ else if (info.inFormat == RTAUDIO_SINT32) {
+ Int32 *in = (Int32 *)inBuffer;
+ for (unsigned int i=0; i<stream_.bufferSize; i++) {
+ for (j=0; j<info.channels; j++) {
+ out[info.outOffset[j]] = (signed char) ((in[info.inOffset[j]] >> 24) & 0x000000ff);
}
+ in += info.inJump;
+ out += info.outJump;
}
- else if (info.inFormat == RTAUDIO_FLOAT32) {
- Float32 *in = (Float32 *)inBuffer;
- for (unsigned int i=0; i<stream_.bufferSize; i++) {
- for (j=0; j<info.channels; j++) {
- out[info.outOffset[j]] = (signed char) (in[info.inOffset[j]] * 127.5 - 0.5);
- }
- in += info.inJump;
- out += info.outJump;
+ }
+ else if (info.inFormat == RTAUDIO_FLOAT32) {
+ Float32 *in = (Float32 *)inBuffer;
+ for (unsigned int i=0; i<stream_.bufferSize; i++) {
+ for (j=0; j<info.channels; j++) {
+ out[info.outOffset[j]] = (signed char) (in[info.inOffset[j]] * 127.5 - 0.5);
}
+ in += info.inJump;
+ out += info.outJump;
}
- else if (info.inFormat == RTAUDIO_FLOAT64) {
- Float64 *in = (Float64 *)inBuffer;
- for (unsigned int i=0; i<stream_.bufferSize; i++) {
- for (j=0; j<info.channels; j++) {
- out[info.outOffset[j]] = (signed char) (in[info.inOffset[j]] * 127.5 - 0.5);
- }
- in += info.inJump;
- out += info.outJump;
+ }
+ else if (info.inFormat == RTAUDIO_FLOAT64) {
+ Float64 *in = (Float64 *)inBuffer;
+ for (unsigned int i=0; i<stream_.bufferSize; i++) {
+ for (j=0; j<info.channels; j++) {
+ out[info.outOffset[j]] = (signed char) (in[info.inOffset[j]] * 127.5 - 0.5);
}
+ in += info.inJump;
+ out += info.outJump;
}
}
}
+}
//static inline uint16_t bswap_16(uint16_t x) { return (x>>8) | (x<<8); }
//static inline uint32_t bswap_32(uint32_t x) { return (bswap_16(x&0xffff)<<16) | (bswap_16(x>>16)); }
//static inline uint64_t bswap_64(uint64_t x) { return (((unsigned long long)bswap_32(x&0xffffffffull))<<32) | (bswap_32(x>>32)); }
- void RtApi :: byteSwapBuffer( char *buffer, unsigned int samples, RtAudioFormat format )
- {
- register char val;
- register char *ptr;
-
- ptr = buffer;
- if ( format == RTAUDIO_SINT16 ) {
- for ( unsigned int i=0; i<samples; i++ ) {
- // Swap 1st and 2nd bytes.
- val = *(ptr);
- *(ptr) = *(ptr+1);
- *(ptr+1) = val;
-
- // Increment 2 bytes.
- ptr += 2;
- }
- }
- else if ( format == RTAUDIO_SINT24 ||
- format == RTAUDIO_SINT32 ||
- format == RTAUDIO_FLOAT32 ) {
- for ( unsigned int i=0; i<samples; i++ ) {
- // Swap 1st and 4th bytes.
- val = *(ptr);
- *(ptr) = *(ptr+3);
- *(ptr+3) = val;
-
- // Swap 2nd and 3rd bytes.
- ptr += 1;
- val = *(ptr);
- *(ptr) = *(ptr+1);
- *(ptr+1) = val;
-
- // Increment 3 more bytes.
- ptr += 3;
- }
- }
- else if ( format == RTAUDIO_FLOAT64 ) {
- for ( unsigned int i=0; i<samples; i++ ) {
- // Swap 1st and 8th bytes
- val = *(ptr);
- *(ptr) = *(ptr+7);
- *(ptr+7) = val;
-
- // Swap 2nd and 7th bytes
- ptr += 1;
- val = *(ptr);
- *(ptr) = *(ptr+5);
- *(ptr+5) = val;
-
- // Swap 3rd and 6th bytes
- ptr += 1;
- val = *(ptr);
- *(ptr) = *(ptr+3);
- *(ptr+3) = val;
-
- // Swap 4th and 5th bytes
- ptr += 1;
- val = *(ptr);
- *(ptr) = *(ptr+1);
- *(ptr+1) = val;
-
- // Increment 5 more bytes.
- ptr += 5;
- }
+void RtApi :: byteSwapBuffer( char *buffer, unsigned int samples, RtAudioFormat format )
+{
+ register char val;
+ register char *ptr;
+
+ ptr = buffer;
+ if ( format == RTAUDIO_SINT16 ) {
+ for ( unsigned int i=0; i<samples; i++ ) {
+ // Swap 1st and 2nd bytes.
+ val = *(ptr);
+ *(ptr) = *(ptr+1);
+ *(ptr+1) = val;
+
+ // Increment 2 bytes.
+ ptr += 2;
+ }
+ }
+ else if ( format == RTAUDIO_SINT24 ||
+ format == RTAUDIO_SINT32 ||
+ format == RTAUDIO_FLOAT32 ) {
+ for ( unsigned int i=0; i<samples; i++ ) {
+ // Swap 1st and 4th bytes.
+ val = *(ptr);
+ *(ptr) = *(ptr+3);
+ *(ptr+3) = val;
+
+ // Swap 2nd and 3rd bytes.
+ ptr += 1;
+ val = *(ptr);
+ *(ptr) = *(ptr+1);
+ *(ptr+1) = val;
+
+ // Increment 3 more bytes.
+ ptr += 3;
+ }
+ }
+ else if ( format == RTAUDIO_FLOAT64 ) {
+ for ( unsigned int i=0; i<samples; i++ ) {
+ // Swap 1st and 8th bytes
+ val = *(ptr);
+ *(ptr) = *(ptr+7);
+ *(ptr+7) = val;
+
+ // Swap 2nd and 7th bytes
+ ptr += 1;
+ val = *(ptr);
+ *(ptr) = *(ptr+5);
+ *(ptr+5) = val;
+
+ // Swap 3rd and 6th bytes
+ ptr += 1;
+ val = *(ptr);
+ *(ptr) = *(ptr+3);
+ *(ptr+3) = val;
+
+ // Swap 4th and 5th bytes
+ ptr += 1;
+ val = *(ptr);
+ *(ptr) = *(ptr+1);
+ *(ptr+1) = val;
+
+ // Increment 5 more bytes.
+ ptr += 5;
}
}
+}
// Indentation settings for Vim and Emacs
//
projects / rtaudio-cdist.git / blobdiff