diff options
Diffstat (limited to 'drivers/rtaudio/RtAudio.cpp')
| -rw-r--r-- | drivers/rtaudio/RtAudio.cpp | 20468 |
1 files changed, 10234 insertions, 10234 deletions
diff --git a/drivers/rtaudio/RtAudio.cpp b/drivers/rtaudio/RtAudio.cpp index 8876f72e21..72ca836907 100644 --- a/drivers/rtaudio/RtAudio.cpp +++ b/drivers/rtaudio/RtAudio.cpp @@ -1,10234 +1,10234 @@ -#ifdef RTAUDIO_ENABLED
-/************************************************************************/
-/*! \class RtAudio
- \brief Realtime audio i/o C++ classes.
-
- RtAudio provides a common API (Application Programming Interface)
- for realtime audio input/output across Linux (native ALSA, Jack,
- and OSS), Macintosh OS X (CoreAudio and Jack), and Windows
- (DirectSound, ASIO and WASAPI) operating systems.
-
- RtAudio WWW site: http://www.music.mcgill.ca/~gary/rtaudio/
-
- RtAudio: realtime audio i/o C++ classes
- Copyright (c) 2001-2014 Gary P. Scavone
-
- Permission is hereby granted, free of charge, to any person
- obtaining a copy of this software and associated documentation files
- (the "Software"), to deal in the Software without restriction,
- including without limitation the rights to use, copy, modify, merge,
- publish, distribute, sublicense, and/or sell copies of the Software,
- and to permit persons to whom the Software is furnished to do so,
- subject to the following conditions:
-
- The above copyright notice and this permission notice shall be
- included in all copies or substantial portions of the Software.
-
- Any person wishing to distribute modifications to the Software is
- asked to send the modifications to the original developer so that
- they can be incorporated into the canonical version. This is,
- however, not a binding provision of this license.
-
- THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND,
- EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF
- MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT.
- IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR
- ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF
- CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION
- WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
-*/
-/************************************************************************/
-
-// RtAudio: Version 4.1.1
-
-#include "RtAudio.h"
-#include <iostream>
-#include <cstdlib>
-#include <cstring>
-#include <climits>
-#include <algorithm>
-
-// Static variable definitions.
-const unsigned int RtApi::MAX_SAMPLE_RATES = 14;
-const unsigned int RtApi::SAMPLE_RATES[] = {
- 4000, 5512, 8000, 9600, 11025, 16000, 22050,
- 32000, 44100, 48000, 88200, 96000, 176400, 192000
-};
-
-#if defined(__WINDOWS_DS__) || defined(__WINDOWS_ASIO__) || defined(__WINDOWS_WASAPI__)
-#ifdef WINRT_ENABLED
- #define MUTEX_INITIALIZE(A) InitializeCriticalSectionEx(A, 0, 0)
-#else
- #define MUTEX_INITIALIZE(A) InitializeCriticalSection(A)
-#endif
- #define MUTEX_DESTROY(A) DeleteCriticalSection(A)
- #define MUTEX_LOCK(A) EnterCriticalSection(A)
- #define MUTEX_UNLOCK(A) LeaveCriticalSection(A)
-
- #include "tchar.h"
-
- static std::string convertCharPointerToStdString(const char *text)
- {
- return std::string(text);
- }
-
- static std::string convertCharPointerToStdString(const wchar_t *text)
- {
- int length = WideCharToMultiByte(CP_UTF8, 0, text, -1, NULL, 0, NULL, NULL);
- std::string s( length-1, '\0' );
- WideCharToMultiByte(CP_UTF8, 0, text, -1, &s[0], length, NULL, NULL);
- return s;
- }
-
-#elif defined(__LINUX_ALSA__) || defined(__LINUX_PULSE__) || defined(__UNIX_JACK__) || defined(__LINUX_OSS__) || defined(__MACOSX_CORE__)
- // pthread API
- #define MUTEX_INITIALIZE(A) pthread_mutex_init(A, NULL)
- #define MUTEX_DESTROY(A) pthread_mutex_destroy(A)
- #define MUTEX_LOCK(A) pthread_mutex_lock(A)
- #define MUTEX_UNLOCK(A) pthread_mutex_unlock(A)
-#else
- #define MUTEX_INITIALIZE(A) abs(*A) // dummy definitions
- #define MUTEX_DESTROY(A) abs(*A) // dummy definitions
-#endif
-
-// *************************************************** //
-//
-// RtAudio definitions.
-//
-// *************************************************** //
-
-std::string RtAudio :: getVersion( void ) throw()
-{
- return RTAUDIO_VERSION;
-}
-
-void RtAudio :: getCompiledApi( std::vector<RtAudio::Api> &apis ) throw()
-{
- apis.clear();
-
- // The order here will control the order of RtAudio's API search in
- // the constructor.
-#if defined(__UNIX_JACK__)
- apis.push_back( UNIX_JACK );
-#endif
-#if defined(__LINUX_ALSA__)
- apis.push_back( LINUX_ALSA );
-#endif
-#if defined(__LINUX_PULSE__)
- apis.push_back( LINUX_PULSE );
-#endif
-#if defined(__LINUX_OSS__)
- apis.push_back( LINUX_OSS );
-#endif
-#if defined(__WINDOWS_ASIO__)
- apis.push_back( WINDOWS_ASIO );
-#endif
-#if defined(__WINDOWS_WASAPI__)
- apis.push_back( WINDOWS_WASAPI );
-#endif
-#if defined(__WINDOWS_DS__)
- apis.push_back( WINDOWS_DS );
-#endif
-#if defined(__MACOSX_CORE__)
- apis.push_back( MACOSX_CORE );
-#endif
-#if defined(__RTAUDIO_DUMMY__)
- apis.push_back( RTAUDIO_DUMMY );
-#endif
-}
-
-void RtAudio :: openRtApi( RtAudio::Api api )
-{
- if ( rtapi_ )
- delete rtapi_;
- rtapi_ = 0;
-
-#if defined(__UNIX_JACK__)
- if ( api == UNIX_JACK )
- rtapi_ = new RtApiJack();
-#endif
-#if defined(__LINUX_ALSA__)
- if ( api == LINUX_ALSA )
- rtapi_ = new RtApiAlsa();
-#endif
-#if defined(__LINUX_PULSE__)
- if ( api == LINUX_PULSE )
- rtapi_ = new RtApiPulse();
-#endif
-#if defined(__LINUX_OSS__)
- if ( api == LINUX_OSS )
- rtapi_ = new RtApiOss();
-#endif
-#if defined(__WINDOWS_ASIO__)
- if ( api == WINDOWS_ASIO )
- rtapi_ = new RtApiAsio();
-#endif
-#if defined(__WINDOWS_WASAPI__)
- if ( api == WINDOWS_WASAPI )
- rtapi_ = new RtApiWasapi();
-#endif
-#if defined(__WINDOWS_DS__)
- if ( api == WINDOWS_DS )
- rtapi_ = new RtApiDs();
-#endif
-#if defined(__MACOSX_CORE__)
- if ( api == MACOSX_CORE )
- rtapi_ = new RtApiCore();
-#endif
-#if defined(__RTAUDIO_DUMMY__)
- if ( api == RTAUDIO_DUMMY )
- rtapi_ = new RtApiDummy();
-#endif
-}
-
-RtAudio :: RtAudio( RtAudio::Api api )
-{
- rtapi_ = 0;
-
- if ( api != UNSPECIFIED ) {
- // Attempt to open the specified API.
- openRtApi( api );
- if ( rtapi_ ) return;
-
- // No compiled support for specified API value. Issue a debug
- // warning and continue as if no API was specified.
- std::cerr << "\nRtAudio: no compiled support for specified API argument!\n" << std::endl;
- }
-
- // Iterate through the compiled APIs and return as soon as we find
- // one with at least one device or we reach the end of the list.
- std::vector< RtAudio::Api > apis;
- getCompiledApi( apis );
- for ( unsigned int i=0; i<apis.size(); i++ ) {
- openRtApi( apis[i] );
- if ( rtapi_ && rtapi_->getDeviceCount() ) break;
- }
-
- if ( rtapi_ ) return;
-
- // It should not be possible to get here because the preprocessor
- // definition __RTAUDIO_DUMMY__ is automatically defined if no
- // API-specific definitions are passed to the compiler. But just in
- // case something weird happens, we'll thow an error.
- std::string errorText = "\nRtAudio: no compiled API support found ... critical error!!\n\n";
- throw( RtAudioError( errorText, RtAudioError::UNSPECIFIED ) );
-}
-
-RtAudio :: ~RtAudio() throw()
-{
- if ( rtapi_ )
- delete rtapi_;
-}
-
-void RtAudio :: openStream( RtAudio::StreamParameters *outputParameters,
- RtAudio::StreamParameters *inputParameters,
- RtAudioFormat format, unsigned int sampleRate,
- unsigned int *bufferFrames,
- RtAudioCallback callback, void *userData,
- RtAudio::StreamOptions *options,
- RtAudioErrorCallback errorCallback )
-{
- return rtapi_->openStream( outputParameters, inputParameters, format,
- sampleRate, bufferFrames, callback,
- userData, options, errorCallback );
-}
-
-// *************************************************** //
-//
-// Public RtApi definitions (see end of file for
-// private or protected utility functions).
-//
-// *************************************************** //
-
-RtApi :: RtApi()
-{
- stream_.state = STREAM_CLOSED;
- stream_.mode = UNINITIALIZED;
- stream_.apiHandle = 0;
- stream_.userBuffer[0] = 0;
- stream_.userBuffer[1] = 0;
- MUTEX_INITIALIZE( &stream_.mutex );
- showWarnings_ = true;
- firstErrorOccurred_ = false;
-}
-
-RtApi :: ~RtApi()
-{
- MUTEX_DESTROY( &stream_.mutex );
-}
-
-void RtApi :: openStream( RtAudio::StreamParameters *oParams,
- RtAudio::StreamParameters *iParams,
- RtAudioFormat format, unsigned int sampleRate,
- unsigned int *bufferFrames,
- RtAudioCallback callback, void *userData,
- RtAudio::StreamOptions *options,
- RtAudioErrorCallback errorCallback )
-{
- if ( stream_.state != STREAM_CLOSED ) {
- errorText_ = "RtApi::openStream: a stream is already open!";
- error( RtAudioError::INVALID_USE );
- return;
- }
-
- // Clear stream information potentially left from a previously open stream.
- clearStreamInfo();
-
- if ( oParams && oParams->nChannels < 1 ) {
- errorText_ = "RtApi::openStream: a non-NULL output StreamParameters structure cannot have an nChannels value less than one.";
- error( RtAudioError::INVALID_USE );
- return;
- }
-
- if ( iParams && iParams->nChannels < 1 ) {
- errorText_ = "RtApi::openStream: a non-NULL input StreamParameters structure cannot have an nChannels value less than one.";
- error( RtAudioError::INVALID_USE );
- return;
- }
-
- if ( oParams == NULL && iParams == NULL ) {
- errorText_ = "RtApi::openStream: input and output StreamParameters structures are both NULL!";
- error( RtAudioError::INVALID_USE );
- return;
- }
-
- if ( formatBytes(format) == 0 ) {
- errorText_ = "RtApi::openStream: 'format' parameter value is undefined.";
- error( RtAudioError::INVALID_USE );
- return;
- }
-
- unsigned int nDevices = getDeviceCount();
- unsigned int oChannels = 0;
- if ( oParams ) {
- oChannels = oParams->nChannels;
- if ( oParams->deviceId >= nDevices ) {
- errorText_ = "RtApi::openStream: output device parameter value is invalid.";
- error( RtAudioError::INVALID_USE );
- return;
- }
- }
-
- unsigned int iChannels = 0;
- if ( iParams ) {
- iChannels = iParams->nChannels;
- if ( iParams->deviceId >= nDevices ) {
- errorText_ = "RtApi::openStream: input device parameter value is invalid.";
- error( RtAudioError::INVALID_USE );
- return;
- }
- }
-
- bool result;
-
- if ( oChannels > 0 ) {
-
- result = probeDeviceOpen( oParams->deviceId, OUTPUT, oChannels, oParams->firstChannel,
- sampleRate, format, bufferFrames, options );
- if ( result == false ) {
- error( RtAudioError::SYSTEM_ERROR );
- return;
- }
- }
-
- if ( iChannels > 0 ) {
-
- result = probeDeviceOpen( iParams->deviceId, INPUT, iChannels, iParams->firstChannel,
- sampleRate, format, bufferFrames, options );
- if ( result == false ) {
- if ( oChannels > 0 ) closeStream();
- error( RtAudioError::SYSTEM_ERROR );
- return;
- }
- }
-
- stream_.callbackInfo.callback = (void *) callback;
- stream_.callbackInfo.userData = userData;
- stream_.callbackInfo.errorCallback = (void *) errorCallback;
-
- if ( options ) options->numberOfBuffers = stream_.nBuffers;
- stream_.state = STREAM_STOPPED;
-}
-
-unsigned int RtApi :: getDefaultInputDevice( void )
-{
- // Should be implemented in subclasses if possible.
- return 0;
-}
-
-unsigned int RtApi :: getDefaultOutputDevice( void )
-{
- // Should be implemented in subclasses if possible.
- return 0;
-}
-
-void RtApi :: closeStream( void )
-{
- // MUST be implemented in subclasses!
- return;
-}
-
-bool RtApi :: probeDeviceOpen( unsigned int /*device*/, StreamMode /*mode*/, unsigned int /*channels*/,
- unsigned int /*firstChannel*/, unsigned int /*sampleRate*/,
- RtAudioFormat /*format*/, unsigned int * /*bufferSize*/,
- RtAudio::StreamOptions * /*options*/ )
-{
- // MUST be implemented in subclasses!
- return FAILURE;
-}
-
-void RtApi :: tickStreamTime( void )
-{
- // Subclasses that do not provide their own implementation of
- // getStreamTime should call this function once per buffer I/O to
- // provide basic stream time support.
-
- stream_.streamTime += ( stream_.bufferSize * 1.0 / stream_.sampleRate );
-
-#if defined( HAVE_GETTIMEOFDAY )
- gettimeofday( &stream_.lastTickTimestamp, NULL );
-#endif
-}
-
-long RtApi :: getStreamLatency( void )
-{
- verifyStream();
-
- long totalLatency = 0;
- if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX )
- totalLatency = stream_.latency[0];
- if ( stream_.mode == INPUT || stream_.mode == DUPLEX )
- totalLatency += stream_.latency[1];
-
- return totalLatency;
-}
-
-double RtApi :: getStreamTime( void )
-{
- verifyStream();
-
-#if defined( HAVE_GETTIMEOFDAY )
- // Return a very accurate estimate of the stream time by
- // adding in the elapsed time since the last tick.
- struct timeval then;
- struct timeval now;
-
- if ( stream_.state != STREAM_RUNNING || stream_.streamTime == 0.0 )
- return stream_.streamTime;
-
- gettimeofday( &now, NULL );
- then = stream_.lastTickTimestamp;
- return stream_.streamTime +
- ((now.tv_sec + 0.000001 * now.tv_usec) -
- (then.tv_sec + 0.000001 * then.tv_usec));
-#else
- return stream_.streamTime;
-#endif
-}
-
-void RtApi :: setStreamTime( double time )
-{
- verifyStream();
-
- if ( time >= 0.0 )
- stream_.streamTime = time;
-}
-
-unsigned int RtApi :: getStreamSampleRate( void )
-{
- verifyStream();
-
- return stream_.sampleRate;
-}
-
-
-// *************************************************** //
-//
-// OS/API-specific methods.
-//
-// *************************************************** //
-
-#if defined(__MACOSX_CORE__)
-
-// The OS X CoreAudio API is designed to use a separate callback
-// procedure for each of its audio devices. A single RtAudio duplex
-// stream using two different devices is supported here, though it
-// cannot be guaranteed to always behave correctly because we cannot
-// synchronize these two callbacks.
-//
-// A property listener is installed for over/underrun information.
-// However, no functionality is currently provided to allow property
-// listeners to trigger user handlers because it is unclear what could
-// be done if a critical stream parameter (buffer size, sample rate,
-// device disconnect) notification arrived. The listeners entail
-// quite a bit of extra code and most likely, a user program wouldn't
-// be prepared for the result anyway. However, we do provide a flag
-// to the client callback function to inform of an over/underrun.
-
-// A structure to hold various information related to the CoreAudio API
-// 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];
- char *deviceBuffer;
- pthread_cond_t condition;
- int drainCounter; // Tracks callback counts when draining
- bool internalDrain; // Indicates if stop is initiated from callback or not.
-
- CoreHandle()
- :deviceBuffer(0), drainCounter(0), internalDrain(false) { nStreams[0] = 1; nStreams[1] = 1; id[0] = 0; id[1] = 0; xrun[0] = false; xrun[1] = false; }
-};
-
-RtApiCore:: RtApiCore()
-{
-#if defined( AVAILABLE_MAC_OS_X_VERSION_10_6_AND_LATER )
- // This is a largely undocumented but absolutely necessary
- // requirement starting with OS-X 10.6. If not called, queries and
- // updates to various audio device properties are not handled
- // correctly.
- CFRunLoopRef theRunLoop = NULL;
- AudioObjectPropertyAddress property = { kAudioHardwarePropertyRunLoop,
- kAudioObjectPropertyScopeGlobal,
- kAudioObjectPropertyElementMaster };
- OSStatus result = AudioObjectSetPropertyData( kAudioObjectSystemObject, &property, 0, NULL, sizeof(CFRunLoopRef), &theRunLoop);
- if ( result != noErr ) {
- errorText_ = "RtApiCore::RtApiCore: error setting run loop property!";
- error( RtAudioError::WARNING );
- }
-#endif
-}
-
-RtApiCore :: ~RtApiCore()
-{
- // The subclass destructor gets called before the base class
- // destructor, so close an existing stream before deallocating
- // apiDeviceId memory.
- if ( stream_.state != STREAM_CLOSED ) closeStream();
-}
-
-unsigned int RtApiCore :: getDeviceCount( void )
-{
- // Find out how many audio devices there are, if any.
- UInt32 dataSize;
- 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( RtAudioError::WARNING );
- return 0;
- }
-
- return dataSize / sizeof( AudioDeviceID );
-}
-
-unsigned int RtApiCore :: getDefaultInputDevice( void )
-{
- unsigned int nDevices = getDeviceCount();
- if ( nDevices <= 1 ) return 0;
-
- AudioDeviceID id;
- UInt32 dataSize = sizeof( AudioDeviceID );
- 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( RtAudioError::WARNING );
- return 0;
- }
-
- dataSize *= nDevices;
- AudioDeviceID deviceList[ nDevices ];
- 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( RtAudioError::WARNING );
- return 0;
- }
-
- for ( unsigned int i=0; i<nDevices; i++ )
- if ( id == deviceList[i] ) return i;
-
- errorText_ = "RtApiCore::getDefaultInputDevice: No default device found!";
- error( RtAudioError::WARNING );
- return 0;
-}
-
-unsigned int RtApiCore :: getDefaultOutputDevice( void )
-{
- unsigned int nDevices = getDeviceCount();
- if ( nDevices <= 1 ) return 0;
-
- AudioDeviceID id;
- UInt32 dataSize = sizeof( AudioDeviceID );
- 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( RtAudioError::WARNING );
- return 0;
- }
-
- dataSize = sizeof( AudioDeviceID ) * nDevices;
- AudioDeviceID deviceList[ nDevices ];
- 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( RtAudioError::WARNING );
- return 0;
- }
-
- for ( unsigned int i=0; i<nDevices; i++ )
- if ( id == deviceList[i] ) return i;
-
- errorText_ = "RtApiCore::getDefaultOutputDevice: No default device found!";
- error( RtAudioError::WARNING );
- return 0;
-}
-
-RtAudio::DeviceInfo RtApiCore :: getDeviceInfo( unsigned int device )
-{
- RtAudio::DeviceInfo info;
- info.probed = false;
-
- // Get device ID
- unsigned int nDevices = getDeviceCount();
- if ( nDevices == 0 ) {
- errorText_ = "RtApiCore::getDeviceInfo: no devices found!";
- error( RtAudioError::INVALID_USE );
- return info;
- }
-
- if ( device >= nDevices ) {
- errorText_ = "RtApiCore::getDeviceInfo: device ID is invalid!";
- error( RtAudioError::INVALID_USE );
- return info;
- }
-
- AudioDeviceID deviceList[ nDevices ];
- UInt32 dataSize = sizeof( AudioDeviceID ) * nDevices;
- 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( RtAudioError::WARNING );
- return info;
- }
-
- AudioDeviceID id = deviceList[ device ];
-
- // Get the device name.
- info.name.erase();
- 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( RtAudioError::WARNING );
- return info;
- }
-
- //const char *mname = CFStringGetCStringPtr( cfname, CFStringGetSystemEncoding() );
- int length = CFStringGetLength(cfname);
- char *mname = (char *)malloc(length * 3 + 1);
-#if defined( UNICODE ) || defined( _UNICODE )
- CFStringGetCString(cfname, mname, length * 3 + 1, kCFStringEncodingUTF8);
-#else
- CFStringGetCString(cfname, mname, length * 3 + 1, CFStringGetSystemEncoding());
-#endif
- info.name.append( (const char *)mname, strlen(mname) );
- info.name.append( ": " );
- CFRelease( cfname );
- free(mname);
-
- 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( RtAudioError::WARNING );
- return info;
- }
-
- //const char *name = CFStringGetCStringPtr( cfname, CFStringGetSystemEncoding() );
- length = CFStringGetLength(cfname);
- char *name = (char *)malloc(length * 3 + 1);
-#if defined( UNICODE ) || defined( _UNICODE )
- CFStringGetCString(cfname, name, length * 3 + 1, kCFStringEncodingUTF8);
-#else
- CFStringGetCString(cfname, name, length * 3 + 1, CFStringGetSystemEncoding());
-#endif
- info.name.append( (const char *)name, strlen(name) );
- CFRelease( cfname );
- free(name);
-
- // Get the output stream "configuration".
- AudioBufferList *bufferList = nil;
- 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( RtAudioError::WARNING );
- return info;
- }
-
- // Allocate the AudioBufferList.
- bufferList = (AudioBufferList *) malloc( dataSize );
- if ( bufferList == NULL ) {
- errorText_ = "RtApiCore::getDeviceInfo: memory error allocating output AudioBufferList.";
- error( RtAudioError::WARNING );
- return info;
- }
-
- 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();
- error( RtAudioError::WARNING );
- return info;
- }
-
- // Get output channel information.
- unsigned int i, nStreams = bufferList->mNumberBuffers;
- for ( i=0; i<nStreams; i++ )
- info.outputChannels += bufferList->mBuffers[i].mNumberChannels;
- free( bufferList );
-
- // Get the input stream "configuration".
- 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( RtAudioError::WARNING );
- return info;
- }
-
- // Allocate the AudioBufferList.
- bufferList = (AudioBufferList *) malloc( dataSize );
- if ( bufferList == NULL ) {
- errorText_ = "RtApiCore::getDeviceInfo: memory error allocating input AudioBufferList.";
- error( RtAudioError::WARNING );
- return info;
- }
-
- 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();
- error( RtAudioError::WARNING );
- return info;
- }
-
- // Get input channel information.
- nStreams = bufferList->mNumberBuffers;
- for ( i=0; i<nStreams; i++ )
- info.inputChannels += bufferList->mBuffers[i].mNumberChannels;
- free( bufferList );
-
- // 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;
-
- // Probe the device sample rates.
- bool isInput = false;
- if ( info.outputChannels == 0 ) isInput = true;
-
- // Determine the supported sample rates.
- 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();
- error( RtAudioError::WARNING );
- return info;
- }
-
- UInt32 nRanges = dataSize / sizeof( AudioValueRange );
- AudioValueRange rangeList[ nRanges ];
- result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, &rangeList );
- if ( result != kAudioHardwareNoError ) {
- errorStream_ << "RtApiCore::getDeviceInfo: system error (" << getErrorCode( result ) << ") getting sample rates.";
- errorText_ = errorStream_.str();
- error( RtAudioError::WARNING );
- return info;
- }
-
- // The sample rate reporting mechanism is a bit of a mystery. It
- // seems that it can either return individual rates or a range of
- // rates. I assume that if the min / max range values are the same,
- // then that represents a single supported rate and if the min / max
- // range values are different, the device supports an arbitrary
- // range of values (though there might be multiple ranges, so we'll
- // use the most conservative range).
- Float64 minimumRate = 1.0, maximumRate = 10000000000.0;
- bool haveValueRange = false;
- info.sampleRates.clear();
- for ( UInt32 i=0; i<nRanges; i++ ) {
- if ( rangeList[i].mMinimum == rangeList[i].mMaximum ) {
- unsigned int tmpSr = (unsigned int) rangeList[i].mMinimum;
- info.sampleRates.push_back( tmpSr );
-
- if ( !info.preferredSampleRate || ( tmpSr <= 48000 && tmpSr > info.preferredSampleRate ) )
- info.preferredSampleRate = tmpSr;
-
- } else {
- haveValueRange = true;
- if ( rangeList[i].mMinimum > minimumRate ) minimumRate = rangeList[i].mMinimum;
- if ( rangeList[i].mMaximum < maximumRate ) maximumRate = rangeList[i].mMaximum;
- }
- }
-
- if ( haveValueRange ) {
- for ( unsigned int k=0; k<MAX_SAMPLE_RATES; k++ ) {
- if ( SAMPLE_RATES[k] >= (unsigned int) minimumRate && SAMPLE_RATES[k] <= (unsigned int) maximumRate ) {
- info.sampleRates.push_back( SAMPLE_RATES[k] );
-
- if ( !info.preferredSampleRate || ( SAMPLE_RATES[k] <= 48000 && SAMPLE_RATES[k] > info.preferredSampleRate ) )
- info.preferredSampleRate = SAMPLE_RATES[k];
- }
- }
- }
-
- // Sort and remove any redundant values
- std::sort( info.sampleRates.begin(), info.sampleRates.end() );
- info.sampleRates.erase( unique( info.sampleRates.begin(), info.sampleRates.end() ), info.sampleRates.end() );
-
- if ( info.sampleRates.size() == 0 ) {
- errorStream_ << "RtApiCore::probeDeviceInfo: No supported sample rates found for device (" << device << ").";
- errorText_ = errorStream_.str();
- error( RtAudioError::WARNING );
- return info;
- }
-
- // CoreAudio always uses 32-bit floating point data for PCM streams.
- // Thus, any other "physical" formats supported by the device are of
- // no interest to the client.
- info.nativeFormats = RTAUDIO_FLOAT32;
-
- 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;
-}
-
-static OSStatus callbackHandler( AudioDeviceID inDevice,
- const AudioTimeStamp* /*inNow*/,
- const AudioBufferList* inInputData,
- const AudioTimeStamp* /*inInputTime*/,
- AudioBufferList* outOutputData,
- const AudioTimeStamp* /*inOutputTime*/,
- void* infoPointer )
-{
- CallbackInfo *info = (CallbackInfo *) infoPointer;
-
- RtApiCore *object = (RtApiCore *) info->object;
- if ( object->callbackEvent( inDevice, inInputData, outOutputData ) == false )
- return kAudioHardwareUnspecifiedError;
- else
- return kAudioHardwareNoError;
-}
-
-static OSStatus xrunListener( AudioObjectID /*inDevice*/,
- UInt32 nAddresses,
- const AudioObjectPropertyAddress properties[],
- void* handlePointer )
-{
- CoreHandle *handle = (CoreHandle *) handlePointer;
- 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 OSStatus rateListener( AudioObjectID inDevice,
- UInt32 /*nAddresses*/,
- const AudioObjectPropertyAddress /*properties*/[],
- void* ratePointer )
-{
- Float64 *rate = (Float64 *) ratePointer;
- UInt32 dataSize = sizeof( Float64 );
- AudioObjectPropertyAddress property = { kAudioDevicePropertyNominalSampleRate,
- kAudioObjectPropertyScopeGlobal,
- kAudioObjectPropertyElementMaster };
- AudioObjectGetPropertyData( inDevice, &property, 0, NULL, &dataSize, rate );
- return kAudioHardwareNoError;
-}
-
-bool RtApiCore :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,
- unsigned int firstChannel, unsigned int sampleRate,
- RtAudioFormat format, unsigned int *bufferSize,
- RtAudio::StreamOptions *options )
-{
- // Get device ID
- unsigned int nDevices = getDeviceCount();
- if ( nDevices == 0 ) {
- // This should not happen because a check is made before this function is called.
- errorText_ = "RtApiCore::probeDeviceOpen: no devices found!";
- return FAILURE;
- }
-
- if ( device >= nDevices ) {
- // This should not happen because a check is made before this function is called.
- errorText_ = "RtApiCore::probeDeviceOpen: device ID is invalid!";
- return FAILURE;
- }
-
- AudioDeviceID deviceList[ nDevices ];
- UInt32 dataSize = sizeof( AudioDeviceID ) * nDevices;
- 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;
- }
-
- AudioDeviceID id = deviceList[ device ];
-
- // Setup for stream mode.
- bool isInput = false;
- if ( mode == INPUT ) {
- isInput = true;
- property.mScope = kAudioDevicePropertyScopeInput;
- }
- else
- property.mScope = kAudioDevicePropertyScopeOutput;
-
- // Get the stream "configuration".
- 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;
- }
-
- // Allocate the AudioBufferList.
- bufferList = (AudioBufferList *) malloc( dataSize );
- if ( bufferList == NULL ) {
- errorText_ = "RtApiCore::probeDeviceOpen: memory error allocating AudioBufferList.";
- return FAILURE;
- }
-
- result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, bufferList );
- if (result != noErr || dataSize == 0) {
- free( bufferList );
- errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting stream configuration for device (" << device << ").";
- errorText_ = errorStream_.str();
- return FAILURE;
- }
-
- // Search for one or more streams that contain the desired number of
- // channels. CoreAudio devices can have an arbitrary number of
- // streams and each stream can have an arbitrary number of channels.
- // For each stream, a single buffer of interleaved samples is
- // provided. RtAudio prefers the use of one stream of interleaved
- // data or multiple consecutive single-channel streams. However, we
- // now support multiple consecutive multi-channel streams of
- // interleaved data as well.
- UInt32 iStream, offsetCounter = firstChannel;
- UInt32 nStreams = bufferList->mNumberBuffers;
- bool monoMode = false;
- bool foundStream = false;
-
- // First check that the device supports the requested number of
- // channels.
- UInt32 deviceChannels = 0;
- for ( iStream=0; iStream<nStreams; iStream++ )
- deviceChannels += bufferList->mBuffers[iStream].mNumberChannels;
-
- if ( deviceChannels < ( channels + firstChannel ) ) {
- free( bufferList );
- errorStream_ << "RtApiCore::probeDeviceOpen: the device (" << device << ") does not support the requested channel count.";
- errorText_ = errorStream_.str();
- return FAILURE;
- }
-
- // Look for a single stream meeting our needs.
- UInt32 firstStream, streamCount = 1, streamChannels = 0, channelOffset = 0;
- for ( iStream=0; iStream<nStreams; iStream++ ) {
- streamChannels = bufferList->mBuffers[iStream].mNumberChannels;
- if ( streamChannels >= channels + offsetCounter ) {
- firstStream = iStream;
- channelOffset = offsetCounter;
- foundStream = true;
- break;
- }
- if ( streamChannels > offsetCounter ) break;
- offsetCounter -= streamChannels;
- }
-
- // If we didn't find a single stream above, then we should be able
- // to meet the channel specification with multiple streams.
- if ( foundStream == false ) {
- monoMode = true;
- offsetCounter = firstChannel;
- for ( iStream=0; iStream<nStreams; iStream++ ) {
- streamChannels = bufferList->mBuffers[iStream].mNumberChannels;
- if ( streamChannels > offsetCounter ) break;
- offsetCounter -= streamChannels;
- }
-
- firstStream = iStream;
- channelOffset = offsetCounter;
- Int32 channelCounter = channels + offsetCounter - streamChannels;
-
- if ( streamChannels > 1 ) monoMode = false;
- while ( channelCounter > 0 ) {
- streamChannels = bufferList->mBuffers[++iStream].mNumberChannels;
- if ( streamChannels > 1 ) monoMode = false;
- channelCounter -= streamChannels;
- streamCount++;
- }
- }
-
- free( bufferList );
-
- // Determine the buffer size.
- AudioValueRange bufferRange;
- dataSize = sizeof( AudioValueRange );
- 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();
- return FAILURE;
- }
-
- if ( bufferRange.mMinimum > *bufferSize ) *bufferSize = (unsigned long) bufferRange.mMinimum;
- else if ( bufferRange.mMaximum < *bufferSize ) *bufferSize = (unsigned long) bufferRange.mMaximum;
- if ( options && options->flags & RTAUDIO_MINIMIZE_LATENCY ) *bufferSize = (unsigned long) bufferRange.mMinimum;
-
- // Set the buffer size. For multiple streams, I'm assuming we only
- // need to make this setting for the master channel.
- UInt32 theSize = (UInt32) *bufferSize;
- dataSize = sizeof( UInt32 );
- 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 << ").";
- errorText_ = errorStream_.str();
- return FAILURE;
- }
-
- // If attempting to setup a duplex stream, the bufferSize parameter
- // MUST be the same in both directions!
- *bufferSize = theSize;
- if ( stream_.mode == OUTPUT && mode == INPUT && *bufferSize != stream_.bufferSize ) {
- errorStream_ << "RtApiCore::probeDeviceOpen: system error setting buffer size for duplex stream on device (" << device << ").";
- errorText_ = errorStream_.str();
- return FAILURE;
- }
-
- stream_.bufferSize = *bufferSize;
- stream_.nBuffers = 1;
-
- // 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;
- }
- }
- }
-
- // 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 ) {
-
- // Set a property listener for the sample rate change
- Float64 reportedRate = 0.0;
- AudioObjectPropertyAddress tmp = { kAudioDevicePropertyNominalSampleRate, kAudioObjectPropertyScopeGlobal, kAudioObjectPropertyElementMaster };
- result = AudioObjectAddPropertyListener( id, &tmp, rateListener, (void *) &reportedRate );
- if ( result != noErr ) {
- errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") setting sample rate property listener for device (" << device << ").";
- errorText_ = errorStream_.str();
- return FAILURE;
- }
-
- nominalRate = (Float64) sampleRate;
- result = AudioObjectSetPropertyData( id, &property, 0, NULL, dataSize, &nominalRate );
- if ( result != noErr ) {
- AudioObjectRemovePropertyListener( id, &tmp, rateListener, (void *) &reportedRate );
- errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") setting sample rate for device (" << device << ").";
- errorText_ = errorStream_.str();
- return FAILURE;
- }
-
- // Now wait until the reported nominal rate is what we just set.
- UInt32 microCounter = 0;
- while ( reportedRate != nominalRate ) {
- microCounter += 5000;
- if ( microCounter > 5000000 ) break;
- usleep( 5000 );
- }
-
- // Remove the property listener.
- AudioObjectRemovePropertyListener( id, &tmp, rateListener, (void *) &reportedRate );
-
- if ( microCounter > 5000000 ) {
- errorStream_ << "RtApiCore::probeDeviceOpen: timeout waiting for sample rate update for device (" << device << ").";
- errorText_ = errorStream_.str();
- return FAILURE;
- }
- }
-
- // Now set the stream format for all streams. Also, check the
- // physical format of the device and change that if necessary.
- AudioStreamBasicDescription description;
- dataSize = sizeof( AudioStreamBasicDescription );
- property.mSelector = kAudioStreamPropertyVirtualFormat;
- result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, &description );
- if ( result != noErr ) {
- errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting stream format for device (" << device << ").";
- errorText_ = errorStream_.str();
- return FAILURE;
- }
-
- // Set the sample rate and data format id. However, only make the
- // change if the sample rate is not within 1.0 of the desired
- // rate and the format is not linear pcm.
- bool updateFormat = false;
- if ( fabs( description.mSampleRate - (Float64)sampleRate ) > 1.0 ) {
- description.mSampleRate = (Float64) sampleRate;
- updateFormat = true;
- }
-
- if ( description.mFormatID != kAudioFormatLinearPCM ) {
- description.mFormatID = kAudioFormatLinearPCM;
- updateFormat = true;
- }
-
- if ( updateFormat ) {
- result = AudioObjectSetPropertyData( id, &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();
- return FAILURE;
- }
- }
-
- // Now check the physical format.
- property.mSelector = kAudioStreamPropertyPhysicalFormat;
- result = AudioObjectGetPropertyData( id, &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();
- return FAILURE;
- }
-
- //std::cout << "Current physical stream format:" << std::endl;
- //std::cout << " mBitsPerChan = " << description.mBitsPerChannel << std::endl;
- //std::cout << " aligned high = " << (description.mFormatFlags & kAudioFormatFlagIsAlignedHigh) << ", isPacked = " << (description.mFormatFlags & kAudioFormatFlagIsPacked) << std::endl;
- //std::cout << " bytesPerFrame = " << description.mBytesPerFrame << std::endl;
- //std::cout << " sample rate = " << description.mSampleRate << std::endl;
-
- if ( description.mFormatID != kAudioFormatLinearPCM || description.mBitsPerChannel < 16 ) {
- description.mFormatID = kAudioFormatLinearPCM;
- //description.mSampleRate = (Float64) sampleRate;
- AudioStreamBasicDescription testDescription = description;
- UInt32 formatFlags;
-
- // We'll try higher bit rates first and then work our way down.
- std::vector< std::pair<UInt32, UInt32> > physicalFormats;
- formatFlags = (description.mFormatFlags | kLinearPCMFormatFlagIsFloat) & ~kLinearPCMFormatFlagIsSignedInteger;
- physicalFormats.push_back( std::pair<Float32, UInt32>( 32, formatFlags ) );
- formatFlags = (description.mFormatFlags | kLinearPCMFormatFlagIsSignedInteger | kAudioFormatFlagIsPacked) & ~kLinearPCMFormatFlagIsFloat;
- physicalFormats.push_back( std::pair<Float32, UInt32>( 32, formatFlags ) );
- physicalFormats.push_back( std::pair<Float32, UInt32>( 24, formatFlags ) ); // 24-bit packed
- formatFlags &= ~( kAudioFormatFlagIsPacked | kAudioFormatFlagIsAlignedHigh );
- physicalFormats.push_back( std::pair<Float32, UInt32>( 24.2, formatFlags ) ); // 24-bit in 4 bytes, aligned low
- formatFlags |= kAudioFormatFlagIsAlignedHigh;
- physicalFormats.push_back( std::pair<Float32, UInt32>( 24.4, formatFlags ) ); // 24-bit in 4 bytes, aligned high
- formatFlags = (description.mFormatFlags | kLinearPCMFormatFlagIsSignedInteger | kAudioFormatFlagIsPacked) & ~kLinearPCMFormatFlagIsFloat;
- physicalFormats.push_back( std::pair<Float32, UInt32>( 16, formatFlags ) );
- physicalFormats.push_back( std::pair<Float32, UInt32>( 8, formatFlags ) );
-
- bool setPhysicalFormat = false;
- for( unsigned int i=0; i<physicalFormats.size(); i++ ) {
- testDescription = description;
- testDescription.mBitsPerChannel = (UInt32) physicalFormats[i].first;
- testDescription.mFormatFlags = physicalFormats[i].second;
- if ( (24 == (UInt32)physicalFormats[i].first) && ~( physicalFormats[i].second & kAudioFormatFlagIsPacked ) )
- testDescription.mBytesPerFrame = 4 * testDescription.mChannelsPerFrame;
- else
- testDescription.mBytesPerFrame = testDescription.mBitsPerChannel/8 * testDescription.mChannelsPerFrame;
- testDescription.mBytesPerPacket = testDescription.mBytesPerFrame * testDescription.mFramesPerPacket;
- result = AudioObjectSetPropertyData( id, &property, 0, NULL, dataSize, &testDescription );
- if ( result == noErr ) {
- setPhysicalFormat = true;
- //std::cout << "Updated physical stream format:" << std::endl;
- //std::cout << " mBitsPerChan = " << testDescription.mBitsPerChannel << std::endl;
- //std::cout << " aligned high = " << (testDescription.mFormatFlags & kAudioFormatFlagIsAlignedHigh) << ", isPacked = " << (testDescription.mFormatFlags & kAudioFormatFlagIsPacked) << std::endl;
- //std::cout << " bytesPerFrame = " << testDescription.mBytesPerFrame << std::endl;
- //std::cout << " sample rate = " << testDescription.mSampleRate << std::endl;
- break;
- }
- }
-
- if ( !setPhysicalFormat ) {
- errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") setting physical data format for device (" << device << ").";
- errorText_ = errorStream_.str();
- return FAILURE;
- }
- } // done setting virtual/physical formats.
-
- // Get the stream / device latency.
- UInt32 latency;
- dataSize = sizeof( UInt32 );
- 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 << ").";
- errorText_ = errorStream_.str();
- error( RtAudioError::WARNING );
- }
- }
-
- // Byte-swapping: According to AudioHardware.h, the stream data will
- // always be presented in native-endian format, so we should never
- // need to byte swap.
- stream_.doByteSwap[mode] = false;
-
- // From the CoreAudio documentation, PCM data must be supplied as
- // 32-bit floats.
- stream_.userFormat = format;
- stream_.deviceFormat[mode] = RTAUDIO_FLOAT32;
-
- if ( streamCount == 1 )
- stream_.nDeviceChannels[mode] = description.mChannelsPerFrame;
- else // multiple streams
- stream_.nDeviceChannels[mode] = channels;
- stream_.nUserChannels[mode] = channels;
- stream_.channelOffset[mode] = channelOffset; // offset within a CoreAudio stream
- if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) stream_.userInterleaved = false;
- else stream_.userInterleaved = true;
- stream_.deviceInterleaved[mode] = true;
- if ( monoMode == true ) 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_.nUserChannels[mode] < stream_.nDeviceChannels[mode] )
- stream_.doConvertBuffer[mode] = true;
- if ( streamCount == 1 ) {
- if ( stream_.nUserChannels[mode] > 1 &&
- stream_.userInterleaved != stream_.deviceInterleaved[mode] )
- stream_.doConvertBuffer[mode] = true;
- }
- else if ( monoMode && stream_.userInterleaved )
- stream_.doConvertBuffer[mode] = true;
-
- // Allocate our CoreHandle structure for the stream.
- CoreHandle *handle = 0;
- if ( stream_.apiHandle == 0 ) {
- try {
- handle = new CoreHandle;
- }
- catch ( std::bad_alloc& ) {
- errorText_ = "RtApiCore::probeDeviceOpen: error allocating CoreHandle memory.";
- goto error;
- }
-
- if ( pthread_cond_init( &handle->condition, NULL ) ) {
- errorText_ = "RtApiCore::probeDeviceOpen: error initializing pthread condition variable.";
- goto error;
- }
- stream_.apiHandle = (void *) handle;
- }
- else
- handle = (CoreHandle *) stream_.apiHandle;
- handle->iStream[mode] = firstStream;
- handle->nStreams[mode] = streamCount;
- handle->id[mode] = id;
-
- // Allocate necessary internal buffers.
- unsigned long bufferBytes;
- bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat );
- // stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 );
- stream_.userBuffer[mode] = (char *) malloc( bufferBytes * sizeof(char) );
- memset( stream_.userBuffer[mode], 0, bufferBytes * sizeof(char) );
- if ( stream_.userBuffer[mode] == NULL ) {
- errorText_ = "RtApiCore::probeDeviceOpen: error allocating user buffer memory.";
- goto error;
- }
-
- // If possible, we will make use of the CoreAudio stream buffers as
- // "device buffers". However, we can't do this if using multiple
- // streams.
- if ( stream_.doConvertBuffer[mode] && handle->nStreams[mode] > 1 ) {
-
- 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_ = "RtApiCore::probeDeviceOpen: error allocating device buffer memory.";
- goto error;
- }
- }
- }
-
- stream_.sampleRate = sampleRate;
- stream_.device[mode] = device;
- stream_.state = STREAM_STOPPED;
- stream_.callbackInfo.object = (void *) this;
-
- // Setup the buffer conversion information structure.
- if ( stream_.doConvertBuffer[mode] ) {
- if ( streamCount > 1 ) setConvertInfo( mode, 0 );
- else setConvertInfo( mode, channelOffset );
- }
-
- if ( mode == INPUT && stream_.mode == OUTPUT && stream_.device[0] == device )
- // Only one callback procedure per device.
- stream_.mode = DUPLEX;
- else {
-#if defined( MAC_OS_X_VERSION_10_5 ) && ( MAC_OS_X_VERSION_MIN_REQUIRED >= MAC_OS_X_VERSION_10_5 )
- result = AudioDeviceCreateIOProcID( id, callbackHandler, (void *) &stream_.callbackInfo, &handle->procId[mode] );
-#else
- // deprecated in favor of AudioDeviceCreateIOProcID()
- result = AudioDeviceAddIOProc( id, callbackHandler, (void *) &stream_.callbackInfo );
-#endif
- if ( result != noErr ) {
- errorStream_ << "RtApiCore::probeDeviceOpen: system error setting callback for device (" << device << ").";
- errorText_ = errorStream_.str();
- goto error;
- }
- if ( stream_.mode == OUTPUT && mode == INPUT )
- stream_.mode = DUPLEX;
- else
- stream_.mode = mode;
- }
-
- // Setup the device property listener for over/underload.
- property.mSelector = kAudioDeviceProcessorOverload;
- property.mScope = kAudioObjectPropertyScopeGlobal;
- result = AudioObjectAddPropertyListener( id, &property, xrunListener, (void *) handle );
-
- return SUCCESS;
-
- error:
- if ( handle ) {
- 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;
- }
- }
-
- if ( stream_.deviceBuffer ) {
- free( stream_.deviceBuffer );
- stream_.deviceBuffer = 0;
- }
-
- stream_.state = STREAM_CLOSED;
- return FAILURE;
-}
-
-void RtApiCore :: closeStream( void )
-{
- if ( stream_.state == STREAM_CLOSED ) {
- errorText_ = "RtApiCore::closeStream(): no open stream to close!";
- error( RtAudioError::WARNING );
- return;
- }
-
- CoreHandle *handle = (CoreHandle *) stream_.apiHandle;
- if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
- if (handle) {
- AudioObjectPropertyAddress property = { kAudioHardwarePropertyDevices,
- kAudioObjectPropertyScopeGlobal,
- kAudioObjectPropertyElementMaster };
-
- property.mSelector = kAudioDeviceProcessorOverload;
- property.mScope = kAudioObjectPropertyScopeGlobal;
- if (AudioObjectRemovePropertyListener( handle->id[0], &property, xrunListener, (void *) handle ) != noErr) {
- errorText_ = "RtApiCore::closeStream(): error removing property listener!";
- error( RtAudioError::WARNING );
- }
- }
- if ( stream_.state == STREAM_RUNNING )
- AudioDeviceStop( handle->id[0], callbackHandler );
-#if defined( MAC_OS_X_VERSION_10_5 ) && ( MAC_OS_X_VERSION_MIN_REQUIRED >= MAC_OS_X_VERSION_10_5 )
- AudioDeviceDestroyIOProcID( handle->id[0], handle->procId[0] );
-#else
- // deprecated in favor of AudioDeviceDestroyIOProcID()
- AudioDeviceRemoveIOProc( handle->id[0], callbackHandler );
-#endif
- }
-
- if ( stream_.mode == INPUT || ( stream_.mode == DUPLEX && stream_.device[0] != stream_.device[1] ) ) {
- if (handle) {
- AudioObjectPropertyAddress property = { kAudioHardwarePropertyDevices,
- kAudioObjectPropertyScopeGlobal,
- kAudioObjectPropertyElementMaster };
-
- property.mSelector = kAudioDeviceProcessorOverload;
- property.mScope = kAudioObjectPropertyScopeGlobal;
- if (AudioObjectRemovePropertyListener( handle->id[1], &property, xrunListener, (void *) handle ) != noErr) {
- errorText_ = "RtApiCore::closeStream(): error removing property listener!";
- error( RtAudioError::WARNING );
- }
- }
- if ( stream_.state == STREAM_RUNNING )
- AudioDeviceStop( handle->id[1], callbackHandler );
-#if defined( MAC_OS_X_VERSION_10_5 ) && ( MAC_OS_X_VERSION_MIN_REQUIRED >= MAC_OS_X_VERSION_10_5 )
- AudioDeviceDestroyIOProcID( handle->id[1], handle->procId[1] );
-#else
- // deprecated in favor of AudioDeviceDestroyIOProcID()
- AudioDeviceRemoveIOProc( handle->id[1], callbackHandler );
-#endif
- }
-
- 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;
- }
-
- // Destroy pthread condition variable.
- pthread_cond_destroy( &handle->condition );
- delete handle;
- stream_.apiHandle = 0;
-
- stream_.mode = UNINITIALIZED;
- stream_.state = STREAM_CLOSED;
-}
-
-void RtApiCore :: startStream( void )
-{
- verifyStream();
- if ( stream_.state == STREAM_RUNNING ) {
- errorText_ = "RtApiCore::startStream(): the stream is already running!";
- error( RtAudioError::WARNING );
- return;
- }
-
- OSStatus result = noErr;
- CoreHandle *handle = (CoreHandle *) stream_.apiHandle;
- if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
-
- result = AudioDeviceStart( handle->id[0], callbackHandler );
- if ( result != noErr ) {
- errorStream_ << "RtApiCore::startStream: system error (" << getErrorCode( result ) << ") starting callback procedure on device (" << stream_.device[0] << ").";
- errorText_ = errorStream_.str();
- goto unlock;
- }
- }
-
- if ( stream_.mode == INPUT ||
- ( stream_.mode == DUPLEX && stream_.device[0] != stream_.device[1] ) ) {
-
- result = AudioDeviceStart( handle->id[1], callbackHandler );
- if ( result != noErr ) {
- errorStream_ << "RtApiCore::startStream: system error starting input callback procedure on device (" << stream_.device[1] << ").";
- errorText_ = errorStream_.str();
- goto unlock;
- }
- }
-
- handle->drainCounter = 0;
- handle->internalDrain = false;
- stream_.state = STREAM_RUNNING;
-
- unlock:
- if ( result == noErr ) return;
- error( RtAudioError::SYSTEM_ERROR );
-}
-
-void RtApiCore :: stopStream( void )
-{
- verifyStream();
- if ( stream_.state == STREAM_STOPPED ) {
- errorText_ = "RtApiCore::stopStream(): the stream is already stopped!";
- error( RtAudioError::WARNING );
- return;
- }
-
- OSStatus result = noErr;
- CoreHandle *handle = (CoreHandle *) stream_.apiHandle;
- if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
-
- if ( handle->drainCounter == 0 ) {
- handle->drainCounter = 2;
- pthread_cond_wait( &handle->condition, &stream_.mutex ); // block until signaled
- }
-
- result = AudioDeviceStop( handle->id[0], callbackHandler );
- if ( result != noErr ) {
- errorStream_ << "RtApiCore::stopStream: system error (" << getErrorCode( result ) << ") stopping callback procedure on device (" << stream_.device[0] << ").";
- errorText_ = errorStream_.str();
- goto unlock;
- }
- }
-
- if ( stream_.mode == INPUT || ( stream_.mode == DUPLEX && stream_.device[0] != stream_.device[1] ) ) {
-
- result = AudioDeviceStop( handle->id[1], callbackHandler );
- if ( result != noErr ) {
- errorStream_ << "RtApiCore::stopStream: system error (" << getErrorCode( result ) << ") stopping input callback procedure on device (" << stream_.device[1] << ").";
- errorText_ = errorStream_.str();
- goto unlock;
- }
- }
-
- stream_.state = STREAM_STOPPED;
-
- unlock:
- if ( result == noErr ) return;
- error( RtAudioError::SYSTEM_ERROR );
-}
-
-void RtApiCore :: abortStream( void )
-{
- verifyStream();
- if ( stream_.state == STREAM_STOPPED ) {
- errorText_ = "RtApiCore::abortStream(): the stream is already stopped!";
- error( RtAudioError::WARNING );
- return;
- }
-
- CoreHandle *handle = (CoreHandle *) stream_.apiHandle;
- handle->drainCounter = 2;
-
- stopStream();
-}
-
-// 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 better to handle it this way because the
-// callbackEvent() function probably should return before the AudioDeviceStop()
-// function is called.
-static void *coreStopStream( void *ptr )
-{
- CallbackInfo *info = (CallbackInfo *) ptr;
- RtApiCore *object = (RtApiCore *) info->object;
-
- object->stopStream();
- pthread_exit( NULL );
-}
-
-bool RtApiCore :: callbackEvent( AudioDeviceID deviceId,
- const AudioBufferList *inBufferList,
- const AudioBufferList *outBufferList )
-{
- if ( stream_.state == STREAM_STOPPED || stream_.state == STREAM_STOPPING ) return SUCCESS;
- if ( stream_.state == STREAM_CLOSED ) {
- errorText_ = "RtApiCore::callbackEvent(): the stream is closed ... this shouldn't happen!";
- error( RtAudioError::WARNING );
- return FAILURE;
- }
-
- CallbackInfo *info = (CallbackInfo *) &stream_.callbackInfo;
- CoreHandle *handle = (CoreHandle *) stream_.apiHandle;
-
- // Check if we were draining the stream and signal is finished.
- if ( handle->drainCounter > 3 ) {
- ThreadHandle threadId;
-
- stream_.state = STREAM_STOPPING;
- if ( handle->internalDrain == true )
- pthread_create( &threadId, NULL, coreStopStream, info );
- else // external call to stopStream()
- pthread_cond_signal( &handle->condition );
- return SUCCESS;
- }
-
- AudioDeviceID outputDevice = handle->id[0];
-
- // Invoke user callback to get fresh output data UNLESS we are
- // draining stream or duplex mode AND the input/output devices are
- // different AND this function is called for the input device.
- if ( handle->drainCounter == 0 && ( stream_.mode != DUPLEX || deviceId == outputDevice ) ) {
- 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;
- }
-
- int cbReturnValue = callback( stream_.userBuffer[0], stream_.userBuffer[1],
- stream_.bufferSize, streamTime, status, info->userData );
- if ( cbReturnValue == 2 ) {
- stream_.state = STREAM_STOPPING;
- handle->drainCounter = 2;
- abortStream();
- return SUCCESS;
- }
- else if ( cbReturnValue == 1 ) {
- handle->drainCounter = 1;
- handle->internalDrain = true;
- }
- }
-
- if ( stream_.mode == OUTPUT || ( stream_.mode == DUPLEX && deviceId == outputDevice ) ) {
-
- if ( handle->drainCounter > 1 ) { // write zeros to the output stream
-
- if ( handle->nStreams[0] == 1 ) {
- memset( outBufferList->mBuffers[handle->iStream[0]].mData,
- 0,
- outBufferList->mBuffers[handle->iStream[0]].mDataByteSize );
- }
- else { // fill multiple streams with zeros
- for ( unsigned int i=0; i<handle->nStreams[0]; i++ ) {
- memset( outBufferList->mBuffers[handle->iStream[0]+i].mData,
- 0,
- outBufferList->mBuffers[handle->iStream[0]+i].mDataByteSize );
- }
- }
- }
- else if ( handle->nStreams[0] == 1 ) {
- if ( stream_.doConvertBuffer[0] ) { // convert directly to CoreAudio stream buffer
- convertBuffer( (char *) outBufferList->mBuffers[handle->iStream[0]].mData,
- stream_.userBuffer[0], stream_.convertInfo[0] );
- }
- else { // copy from user buffer
- memcpy( outBufferList->mBuffers[handle->iStream[0]].mData,
- stream_.userBuffer[0],
- outBufferList->mBuffers[handle->iStream[0]].mDataByteSize );
- }
- }
- else { // fill multiple streams
- Float32 *inBuffer = (Float32 *) stream_.userBuffer[0];
- if ( stream_.doConvertBuffer[0] ) {
- convertBuffer( stream_.deviceBuffer, stream_.userBuffer[0], stream_.convertInfo[0] );
- inBuffer = (Float32 *) stream_.deviceBuffer;
- }
-
- if ( stream_.deviceInterleaved[0] == false ) { // mono mode
- 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,
- (void *)&inBuffer[i*stream_.bufferSize], bufferBytes );
- }
- }
- else { // fill multiple multi-channel streams with interleaved data
- UInt32 streamChannels, channelsLeft, inJump, outJump, inOffset;
- Float32 *out, *in;
-
- 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 and not mono mode
- inChannels = stream_.nDeviceChannels[0];
- }
-
- if ( inInterleaved ) inOffset = 1;
- else inOffset = stream_.bufferSize;
-
- channelsLeft = inChannels;
- for ( unsigned int i=0; i<handle->nStreams[0]; i++ ) {
- in = inBuffer;
- out = (Float32 *) outBufferList->mBuffers[handle->iStream[0]+i].mData;
- streamChannels = outBufferList->mBuffers[handle->iStream[0]+i].mNumberChannels;
-
- outJump = 0;
- // Account for possible channel offset in first stream
- if ( i == 0 && stream_.channelOffset[0] > 0 ) {
- streamChannels -= stream_.channelOffset[0];
- outJump = stream_.channelOffset[0];
- out += outJump;
- }
-
- // Account for possible unfilled channels at end of the last stream
- if ( streamChannels > channelsLeft ) {
- outJump = streamChannels - channelsLeft;
- streamChannels = channelsLeft;
- }
-
- // Determine input buffer offsets and skips
- if ( inInterleaved ) {
- inJump = inChannels;
- in += inChannels - channelsLeft;
- }
- else {
- inJump = 1;
- in += (inChannels - channelsLeft) * inOffset;
- }
-
- for ( unsigned int i=0; i<stream_.bufferSize; i++ ) {
- for ( unsigned int j=0; j<streamChannels; j++ ) {
- *out++ = in[j*inOffset];
- }
- out += outJump;
- in += inJump;
- }
- channelsLeft -= streamChannels;
- }
- }
- }
- }
-
- // Don't bother draining input
- if ( handle->drainCounter ) {
- handle->drainCounter++;
- goto unlock;
- }
-
- AudioDeviceID inputDevice;
- inputDevice = handle->id[1];
- if ( stream_.mode == INPUT || ( stream_.mode == DUPLEX && deviceId == inputDevice ) ) {
-
- if ( handle->nStreams[1] == 1 ) {
- if ( stream_.doConvertBuffer[1] ) { // convert directly from CoreAudio stream buffer
- convertBuffer( stream_.userBuffer[1],
- (char *) inBufferList->mBuffers[handle->iStream[1]].mData,
- stream_.convertInfo[1] );
- }
- else { // copy to user buffer
- memcpy( stream_.userBuffer[1],
- inBufferList->mBuffers[handle->iStream[1]].mData,
- inBufferList->mBuffers[handle->iStream[1]].mDataByteSize );
- }
- }
- else { // read from multiple streams
- Float32 *outBuffer = (Float32 *) stream_.userBuffer[1];
- if ( stream_.doConvertBuffer[1] ) outBuffer = (Float32 *) stream_.deviceBuffer;
-
- 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( (void *)&outBuffer[i*stream_.bufferSize],
- inBufferList->mBuffers[handle->iStream[1]+i].mData, bufferBytes );
- }
- }
- else { // read from multiple multi-channel streams
- UInt32 streamChannels, channelsLeft, inJump, outJump, outOffset;
- Float32 *out, *in;
-
- 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 and not mono mode
- outChannels = stream_.nDeviceChannels[1];
- }
-
- if ( outInterleaved ) outOffset = 1;
- else outOffset = stream_.bufferSize;
-
- channelsLeft = outChannels;
- for ( unsigned int i=0; i<handle->nStreams[1]; i++ ) {
- out = outBuffer;
- in = (Float32 *) inBufferList->mBuffers[handle->iStream[1]+i].mData;
- streamChannels = inBufferList->mBuffers[handle->iStream[1]+i].mNumberChannels;
-
- inJump = 0;
- // Account for possible channel offset in first stream
- if ( i == 0 && stream_.channelOffset[1] > 0 ) {
- streamChannels -= stream_.channelOffset[1];
- inJump = stream_.channelOffset[1];
- in += inJump;
- }
-
- // Account for possible unread channels at end of the last stream
- if ( streamChannels > channelsLeft ) {
- inJump = streamChannels - channelsLeft;
- streamChannels = channelsLeft;
- }
-
- // Determine output buffer offsets and skips
- if ( outInterleaved ) {
- outJump = outChannels;
- out += outChannels - channelsLeft;
- }
- else {
- outJump = 1;
- out += (outChannels - channelsLeft) * outOffset;
- }
-
- for ( unsigned int i=0; i<stream_.bufferSize; i++ ) {
- for ( unsigned int j=0; j<streamChannels; j++ ) {
- out[j*outOffset] = *in++;
- }
- out += outJump;
- in += inJump;
- }
- channelsLeft -= streamChannels;
- }
- }
-
- if ( stream_.doConvertBuffer[1] ) { // convert from our internal "device" buffer
- convertBuffer( stream_.userBuffer[1],
- stream_.deviceBuffer,
- stream_.convertInfo[1] );
- }
- }
- }
-
- unlock:
- //MUTEX_UNLOCK( &stream_.mutex );
-
- RtApi::tickStreamTime();
- return SUCCESS;
-}
-
-const char* RtApiCore :: getErrorCode( OSStatus code )
-{
- switch( code ) {
-
- case kAudioHardwareNotRunningError:
- return "kAudioHardwareNotRunningError";
-
- case kAudioHardwareUnspecifiedError:
- return "kAudioHardwareUnspecifiedError";
-
- case kAudioHardwareUnknownPropertyError:
- return "kAudioHardwareUnknownPropertyError";
-
- case kAudioHardwareBadPropertySizeError:
- return "kAudioHardwareBadPropertySizeError";
-
- case kAudioHardwareIllegalOperationError:
- return "kAudioHardwareIllegalOperationError";
-
- case kAudioHardwareBadObjectError:
- return "kAudioHardwareBadObjectError";
-
- case kAudioHardwareBadDeviceError:
- return "kAudioHardwareBadDeviceError";
-
- case kAudioHardwareBadStreamError:
- return "kAudioHardwareBadStreamError";
-
- case kAudioHardwareUnsupportedOperationError:
- return "kAudioHardwareUnsupportedOperationError";
-
- case kAudioDeviceUnsupportedFormatError:
- return "kAudioDeviceUnsupportedFormatError";
-
- case kAudioDevicePermissionsError:
- return "kAudioDevicePermissionsError";
-
- 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.
-
-#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; }
-};
-
-static void jackSilentError( const char * ) {};
-
-RtApiJack :: RtApiJack()
-{
- // Nothing to do here.
-#if !defined(__RTAUDIO_DEBUG__)
- // Turn off Jack's internal error reporting.
- jack_set_error_function( &jackSilentError );
-#endif
-}
-
-RtApiJack :: ~RtApiJack()
-{
- if ( stream_.state != STREAM_CLOSED ) closeStream();
-}
-
-unsigned int RtApiJack :: getDeviceCount( void )
-{
- // See if we can become a jack client.
- jack_options_t options = (jack_options_t) ( JackNoStartServer ); //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 );
- }
-
- jack_client_close( client );
- return nDevices;
-}
-
-RtAudio::DeviceInfo RtApiJack :: getDeviceInfo( unsigned int device )
-{
- RtAudio::DeviceInfo info;
- info.probed = false;
-
- jack_options_t options = (jack_options_t) ( JackNoStartServer ); //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( RtAudioError::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 >= nDevices ) {
- jack_client_close( client );
- errorText_ = "RtApiJack::getDeviceInfo: device ID is invalid!";
- error( RtAudioError::INVALID_USE );
- return info;
- }
-
- // Get the current jack server sample rate.
- info.sampleRates.clear();
-
- info.preferredSampleRate = jack_get_sample_rate( client );
- info.sampleRates.push_back( info.preferredSampleRate );
-
- // 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 "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);
- errorText_ = "RtApiJack::getDeviceInfo: error determining Jack input/output channels!";
- error( RtAudioError::WARNING );
- return info;
- }
-
- // 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;
-
- // Jack always uses 32-bit floats.
- info.nativeFormats = RTAUDIO_FLOAT32;
-
- // 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_client_close(client);
- info.probed = true;
- return info;
-}
-
-static int jackCallbackHandler( jack_nframes_t nframes, void *infoPointer )
-{
- CallbackInfo *info = (CallbackInfo *) infoPointer;
-
- RtApiJack *object = (RtApiJack *) info->object;
- if ( object->callbackEvent( (unsigned long) nframes ) == false ) return 1;
-
- return 0;
-}
-
-// 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.
-static void *jackCloseStream( void *ptr )
-{
- CallbackInfo *info = (CallbackInfo *) ptr;
- RtApiJack *object = (RtApiJack *) info->object;
-
- object->closeStream();
-
- pthread_exit( NULL );
-}
-static void jackShutdown( void *infoPointer )
-{
- CallbackInfo *info = (CallbackInfo *) infoPointer;
- RtApiJack *object = (RtApiJack *) info->object;
-
- // 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;
-
- ThreadHandle threadId;
- pthread_create( &threadId, NULL, jackCloseStream, info );
- std::cerr << "\nRtApiJack: the Jack server is shutting down this client ... stream stopped and closed!!\n" << std::endl;
-}
-
-static int jackXrun( void *infoPointer )
-{
- JackHandle *handle = (JackHandle *) infoPointer;
-
- 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 ); //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( RtAudioError::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 );
- }
-
- if ( device >= nDevices ) {
- errorText_ = "RtApiJack::probeDeviceOpen: device ID is invalid!";
- return FAILURE;
- }
-
- // 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 );
- }
-
- // 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;
- }
-
- // 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;
-
- // Get the latency of the JACK port.
- ports = jack_get_ports( client, deviceName.c_str(), NULL, flag );
- if ( ports[ firstChannel ] ) {
- // Added by Ge Wang
- jack_latency_callback_mode_t cbmode = (mode == INPUT ? JackCaptureLatency : JackPlaybackLatency);
- // the range (usually the min and max are equal)
- jack_latency_range_t latrange; latrange.min = latrange.max = 0;
- // get the latency range
- jack_port_get_latency_range( jack_port_by_name( client, ports[firstChannel] ), cbmode, &latrange );
- // be optimistic, use the min!
- stream_.latency[mode] = latrange.min;
- //stream_.latency[mode] = jack_port_get_latency( jack_port_by_name( client, ports[ firstChannel ] ) );
- }
- free( ports );
-
- // The jack server always uses 32-bit floating-point data.
- stream_.deviceFormat[mode] = RTAUDIO_FLOAT32;
- stream_.userFormat = format;
-
- if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) stream_.userInterleaved = false;
- else stream_.userInterleaved = true;
-
- // Jack always uses non-interleaved buffers.
- stream_.deviceInterleaved[mode] = false;
-
- // Jack always provides host byte-ordered data.
- stream_.doByteSwap[mode] = false;
-
- // 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;
-
- 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;
- }
-
- 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;
-
- // 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;
- }
-
- if ( stream_.doConvertBuffer[mode] ) {
-
- 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 ( 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;
- }
- }
- }
-
- // 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;
- }
-
- stream_.device[mode] = device;
- stream_.channelOffset[mode] = firstChannel;
- stream_.state = STREAM_STOPPED;
- stream_.callbackInfo.object = (void *) this;
-
- 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 );
- }
- }
-
- // 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 ( 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] );
-
- delete handle;
- 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 RtApiJack :: closeStream( void )
-{
- if ( stream_.state == STREAM_CLOSED ) {
- errorText_ = "RtApiJack::closeStream(): no open stream to close!";
- error( RtAudioError::WARNING );
- return;
- }
-
- JackHandle *handle = (JackHandle *) stream_.apiHandle;
- if ( handle ) {
-
- if ( stream_.state == STREAM_RUNNING )
- jack_deactivate( 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;
- }
-
- 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;
- }
-
- 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( RtAudioError::WARNING );
- return;
- }
-
- JackHandle *handle = (JackHandle *) stream_.apiHandle;
- int result = jack_activate( handle->client );
- if ( result ) {
- errorText_ = "RtApiJack::startStream(): unable to activate JACK client!";
- goto unlock;
- }
-
- 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;
- }
-
- // 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;
- }
-
- // 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:
- if ( result == 0 ) return;
- error( RtAudioError::SYSTEM_ERROR );
-}
-
-void RtApiJack :: stopStream( void )
-{
- verifyStream();
- if ( stream_.state == STREAM_STOPPED ) {
- errorText_ = "RtApiJack::stopStream(): the stream is already stopped!";
- error( RtAudioError::WARNING );
- return;
- }
-
- JackHandle *handle = (JackHandle *) stream_.apiHandle;
- if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
-
- if ( handle->drainCounter == 0 ) {
- handle->drainCounter = 2;
- pthread_cond_wait( &handle->condition, &stream_.mutex ); // block until signaled
- }
- }
-
- jack_deactivate( handle->client );
- stream_.state = STREAM_STOPPED;
-}
-
-void RtApiJack :: abortStream( void )
-{
- verifyStream();
- if ( stream_.state == STREAM_STOPPED ) {
- errorText_ = "RtApiJack::abortStream(): the stream is already stopped!";
- error( RtAudioError::WARNING );
- return;
- }
-
- JackHandle *handle = (JackHandle *) stream_.apiHandle;
- handle->drainCounter = 2;
-
- stopStream();
-}
-
-// 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.
-static void *jackStopStream( void *ptr )
-{
- CallbackInfo *info = (CallbackInfo *) ptr;
- RtApiJack *object = (RtApiJack *) info->object;
-
- object->stopStream();
- pthread_exit( NULL );
-}
-
-bool RtApiJack :: callbackEvent( unsigned long nframes )
-{
- if ( stream_.state == STREAM_STOPPED || stream_.state == STREAM_STOPPING ) return SUCCESS;
- if ( stream_.state == STREAM_CLOSED ) {
- errorText_ = "RtApiCore::callbackEvent(): the stream is closed ... this shouldn't happen!";
- error( RtAudioError::WARNING );
- return FAILURE;
- }
- if ( stream_.bufferSize != nframes ) {
- errorText_ = "RtApiCore::callbackEvent(): the JACK buffer size has changed ... cannot process!";
- error( RtAudioError::WARNING );
- return FAILURE;
- }
-
- CallbackInfo *info = (CallbackInfo *) &stream_.callbackInfo;
- JackHandle *handle = (JackHandle *) stream_.apiHandle;
-
- // Check if we were draining the stream and signal is finished.
- if ( handle->drainCounter > 3 ) {
- ThreadHandle threadId;
-
- stream_.state = STREAM_STOPPING;
- if ( handle->internalDrain == true )
- pthread_create( &threadId, NULL, jackStopStream, info );
- else
- pthread_cond_signal( &handle->condition );
- return SUCCESS;
- }
-
- // 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;
- }
- int cbReturnValue = callback( stream_.userBuffer[0], stream_.userBuffer[1],
- stream_.bufferSize, streamTime, status, info->userData );
- if ( cbReturnValue == 2 ) {
- stream_.state = STREAM_STOPPING;
- handle->drainCounter = 2;
- ThreadHandle id;
- pthread_create( &id, NULL, jackStopStream, info );
- return SUCCESS;
- }
- else if ( cbReturnValue == 1 ) {
- handle->drainCounter = 1;
- handle->internalDrain = true;
- }
- }
-
- jack_default_audio_sample_t *jackbuffer;
- unsigned long bufferBytes = nframes * sizeof( jack_default_audio_sample_t );
- if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
-
- if ( handle->drainCounter > 1 ) { // write zeros to the output stream
-
- 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 );
- }
-
- }
- else if ( stream_.doConvertBuffer[0] ) {
-
- convertBuffer( stream_.deviceBuffer, stream_.userBuffer[0], stream_.convertInfo[0] );
-
- 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 );
- }
- }
- }
-
- // Don't bother draining input
- if ( handle->drainCounter ) {
- handle->drainCounter++;
- goto unlock;
- }
-
- if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {
-
- 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 );
- }
- }
- }
-
- unlock:
- RtApi::tickStreamTime();
- return SUCCESS;
-}
- //******************** End of __UNIX_JACK__ *********************//
-#endif
-
-#if defined(__WINDOWS_ASIO__) // ASIO API on Windows
-
-// 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.
-
-#include "asiosys.h"
-#include "asio.h"
-#include "iasiothiscallresolver.h"
-#include "asiodrivers.h"
-#include <cmath>
-
-static AsioDrivers drivers;
-static ASIOCallbacks asioCallbacks;
-static ASIODriverInfo driverInfo;
-static CallbackInfo *asioCallbackInfo;
-static bool asioXRun;
-
-struct AsioHandle {
- int drainCounter; // Tracks callback counts when draining
- bool internalDrain; // Indicates if stop is initiated from callback or not.
- ASIOBufferInfo *bufferInfos;
- HANDLE condition;
-
- AsioHandle()
- :drainCounter(0), internalDrain(false), bufferInfos(0) {}
-};
-
-// Function declarations (definitions at end of section)
-static const char* getAsioErrorString( ASIOError result );
-static void sampleRateChanged( ASIOSampleRate sRate );
-static long asioMessages( long selector, long value, void* message, double* opt );
-
-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( RtAudioError::WARNING );
- }
- coInitialized_ = true;
-
- drivers.removeCurrentDriver();
- driverInfo.asioVersion = 2;
-
- // See note in DirectSound implementation about GetDesktopWindow().
- driverInfo.sysRef = GetForegroundWindow();
-}
-
-RtApiAsio :: ~RtApiAsio()
-{
- if ( stream_.state != STREAM_CLOSED ) closeStream();
- if ( coInitialized_ ) CoUninitialize();
-}
-
-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( RtAudioError::INVALID_USE );
- return info;
- }
-
- if ( device >= nDevices ) {
- errorText_ = "RtApiAsio::getDeviceInfo: device ID is invalid!";
- error( RtAudioError::INVALID_USE );
- return info;
- }
-
- // 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( RtAudioError::WARNING );
- return info;
- }
- return devices_[ device ];
- }
-
- 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( RtAudioError::WARNING );
- return info;
- }
-
- info.name = driverName;
-
- if ( !drivers.loadDriver( driverName ) ) {
- errorStream_ << "RtApiAsio::getDeviceInfo: unable to load driver (" << driverName << ").";
- errorText_ = errorStream_.str();
- error( RtAudioError::WARNING );
- return info;
- }
-
- result = ASIOInit( &driverInfo );
- if ( result != ASE_OK ) {
- errorStream_ << "RtApiAsio::getDeviceInfo: error (" << getAsioErrorString( result ) << ") initializing driver (" << driverName << ").";
- errorText_ = errorStream_.str();
- error( RtAudioError::WARNING );
- return info;
- }
-
- // 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( RtAudioError::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;
-
- // 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] );
-
- if ( !info.preferredSampleRate || ( SAMPLE_RATES[i] <= 48000 && SAMPLE_RATES[i] > info.preferredSampleRate ) )
- info.preferredSampleRate = 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( RtAudioError::WARNING );
- return info;
- }
-
- 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;
- else if ( channelInfo.type == ASIOSTInt24MSB || channelInfo.type == ASIOSTInt24LSB )
- info.nativeFormats |= RTAUDIO_SINT24;
-
- if ( info.outputChannels > 0 )
- if ( getDefaultOutputDevice() == device ) info.isDefaultOutput = true;
- if ( info.inputChannels > 0 )
- if ( getDefaultInputDevice() == device ) info.isDefaultInput = true;
-
- info.probed = true;
- drivers.removeCurrentDriver();
- return info;
-}
-
-static void bufferSwitch( long index, ASIOBool /*processNow*/ )
-{
- RtApiAsio *object = (RtApiAsio *) asioCallbackInfo->object;
- object->callbackEvent( index );
-}
-
-void RtApiAsio :: saveDeviceInfo( void )
-{
- devices_.clear();
-
- unsigned int nDevices = getDeviceCount();
- devices_.resize( nDevices );
- for ( unsigned int i=0; i<nDevices; i++ )
- devices_[i] = getDeviceInfo( i );
-}
-
-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 )
-{////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////
-
- bool isDuplexInput = mode == INPUT && stream_.mode == OUTPUT;
-
- // For ASIO, a duplex stream MUST use the same driver.
- if ( isDuplexInput && stream_.device[0] != device ) {
- errorText_ = "RtApiAsio::probeDeviceOpen: an ASIO duplex stream must use the same device for input and output!";
- return FAILURE;
- }
-
- 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;
- }
-
- // Only load the driver once for duplex stream.
- if ( !isDuplexInput ) {
- // 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();
-
- if ( !drivers.loadDriver( driverName ) ) {
- errorStream_ << "RtApiAsio::probeDeviceOpen: unable to load driver (" << driverName << ").";
- errorText_ = errorStream_.str();
- return FAILURE;
- }
-
- result = ASIOInit( &driverInfo );
- if ( result != ASE_OK ) {
- errorStream_ << "RtApiAsio::probeDeviceOpen: error (" << getAsioErrorString( result ) << ") initializing driver (" << driverName << ").";
- errorText_ = errorStream_.str();
- return FAILURE;
- }
- }
-
- // keep them before any "goto error", they are used for error cleanup + goto device boundary checks
- bool buffersAllocated = false;
- AsioHandle *handle = (AsioHandle *) stream_.apiHandle;
- unsigned int nChannels;
-
-
- // Check the device channel count.
- long inputChannels, outputChannels;
- result = ASIOGetChannels( &inputChannels, &outputChannels );
- if ( result != ASE_OK ) {
- errorStream_ << "RtApiAsio::probeDeviceOpen: error (" << getAsioErrorString( result ) << ") getting channel count (" << driverName << ").";
- errorText_ = errorStream_.str();
- goto error;
- }
-
- if ( ( mode == OUTPUT && (channels+firstChannel) > (unsigned int) outputChannels) ||
- ( mode == INPUT && (channels+firstChannel) > (unsigned int) inputChannels) ) {
- errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") does not support requested channel count (" << channels << ") + offset (" << firstChannel << ").";
- errorText_ = errorStream_.str();
- goto error;
- }
- stream_.nDeviceChannels[mode] = channels;
- stream_.nUserChannels[mode] = channels;
- stream_.channelOffset[mode] = firstChannel;
-
- // Verify the sample rate is supported.
- result = ASIOCanSampleRate( (ASIOSampleRate) sampleRate );
- if ( result != ASE_OK ) {
- errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") does not support requested sample rate (" << sampleRate << ").";
- errorText_ = errorStream_.str();
- goto error;
- }
-
- // Get the current sample rate
- ASIOSampleRate currentRate;
- result = ASIOGetSampleRate( ¤tRate );
- if ( result != ASE_OK ) {
- errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error getting sample rate.";
- errorText_ = errorStream_.str();
- goto error;
- }
-
- // Set the sample rate only if necessary
- if ( currentRate != sampleRate ) {
- result = ASIOSetSampleRate( (ASIOSampleRate) sampleRate );
- if ( result != ASE_OK ) {
- errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error setting sample rate (" << sampleRate << ").";
- errorText_ = errorStream_.str();
- goto error;
- }
- }
-
- // 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 ) {
- errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error (" << getAsioErrorString( result ) << ") getting data format.";
- errorText_ = errorStream_.str();
- goto error;
- }
-
- // 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;
- }
- else if ( channelInfo.type == ASIOSTInt24MSB || channelInfo.type == ASIOSTInt24LSB ) {
- stream_.deviceFormat[mode] = RTAUDIO_SINT24;
- if ( channelInfo.type == ASIOSTInt24MSB ) stream_.doByteSwap[mode] = true;
- }
-
- if ( stream_.deviceFormat[mode] == 0 ) {
- errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") data format not supported by RtAudio.";
- errorText_ = errorStream_.str();
- goto error;
- }
-
- // 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 ) {
- errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error (" << getAsioErrorString( result ) << ") getting buffer size.";
- errorText_ = errorStream_.str();
- goto error;
- }
-
- if ( isDuplexInput ) {
- // When this is the duplex input (output was opened before), then we have to use the same
- // buffersize as the output, because it might use the preferred buffer size, which most
- // likely wasn't passed as input to this. The buffer sizes have to be identically anyway,
- // So instead of throwing an error, make them equal. The caller uses the reference
- // to the "bufferSize" param as usual to set up processing buffers.
-
- *bufferSize = stream_.bufferSize;
-
- } else {
- if ( *bufferSize == 0 ) *bufferSize = preferSize;
- else 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;
-
- 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;
- }
- }
-
- /*
- // we don't use it anymore, see above!
- // Just left it here for the case...
- if ( isDuplexInput && stream_.bufferSize != *bufferSize ) {
- errorText_ = "RtApiAsio::probeDeviceOpen: input/output buffersize discrepancy!";
- goto error;
- }
- */
-
- stream_.bufferSize = *bufferSize;
- stream_.nBuffers = 2;
-
- if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) stream_.userInterleaved = false;
- else stream_.userInterleaved = true;
-
- // ASIO always uses non-interleaved buffers.
- stream_.deviceInterleaved[mode] = false;
-
- // Allocate, if necessary, our AsioHandle structure for the stream.
- if ( handle == 0 ) {
- try {
- handle = new AsioHandle;
- }
- catch ( std::bad_alloc& ) {
- errorText_ = "RtApiAsio::probeDeviceOpen: error allocating AsioHandle memory.";
- goto error;
- }
- 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.
- if ( mode == INPUT && stream_.mode == OUTPUT ) {
- ASIODisposeBuffers();
- if ( handle->bufferInfos ) free( handle->bufferInfos );
- }
-
- // Allocate, initialize, and save the bufferInfos in our stream callbackInfo structure.
- 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;
- }
-
- 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;
- }
-
- // prepare for callbacks
- stream_.sampleRate = sampleRate;
- stream_.device[mode] = device;
- stream_.mode = isDuplexInput ? DUPLEX : mode;
-
- // store this class instance before registering callbacks, that are going to use it
- asioCallbackInfo = &stream_.callbackInfo;
- stream_.callbackInfo.object = (void *) this;
-
- // 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 ) {
- // Standard method failed. This can happen with strict/misbehaving drivers that return valid buffer size ranges
- // but only accept the preferred buffer size as parameter for ASIOCreateBuffers. eg. Creatives ASIO driver
- // in that case, let's be naïve and try that instead
- *bufferSize = preferSize;
- stream_.bufferSize = *bufferSize;
- 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;
- stream_.state = STREAM_STOPPED;
-
- // 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;
- }
-
- if ( stream_.doConvertBuffer[mode] ) {
-
- bool makeBuffer = true;
- bufferBytes = stream_.nDeviceChannels[mode] * formatBytes( stream_.deviceFormat[mode] );
- if ( isDuplexInput && 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_ = "RtApiAsio::probeDeviceOpen: error allocating device buffer memory.";
- goto error;
- }
- }
- }
-
- // Determine device latencies
- long inputLatency, outputLatency;
- result = ASIOGetLatencies( &inputLatency, &outputLatency );
- if ( result != ASE_OK ) {
- errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error (" << getAsioErrorString( result ) << ") getting latency.";
- errorText_ = errorStream_.str();
- error( RtAudioError::WARNING); // warn but don't fail
- }
- else {
- stream_.latency[0] = outputLatency;
- stream_.latency[1] = inputLatency;
- }
-
- // 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 ( !isDuplexInput ) {
- // the cleanup for error in the duplex input, is done by RtApi::openStream
- // So we clean up for single channel only
-
- if ( buffersAllocated )
- ASIODisposeBuffers();
-
- drivers.removeCurrentDriver();
-
- if ( handle ) {
- CloseHandle( handle->condition );
- if ( handle->bufferInfos )
- free( handle->bufferInfos );
-
- delete handle;
- stream_.apiHandle = 0;
- }
-
-
- if ( stream_.userBuffer[mode] ) {
- free( stream_.userBuffer[mode] );
- stream_.userBuffer[mode] = 0;
- }
-
- if ( stream_.deviceBuffer ) {
- free( stream_.deviceBuffer );
- stream_.deviceBuffer = 0;
- }
- }
-
- return FAILURE;
-}////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////
-
-void RtApiAsio :: closeStream()
-{
- if ( stream_.state == STREAM_CLOSED ) {
- errorText_ = "RtApiAsio::closeStream(): no open stream to close!";
- error( RtAudioError::WARNING );
- return;
- }
-
- if ( stream_.state == STREAM_RUNNING ) {
- stream_.state = STREAM_STOPPED;
- ASIOStop();
- }
- ASIODisposeBuffers();
- drivers.removeCurrentDriver();
-
- AsioHandle *handle = (AsioHandle *) stream_.apiHandle;
- if ( handle ) {
- CloseHandle( handle->condition );
- if ( handle->bufferInfos )
- free( handle->bufferInfos );
- delete handle;
- 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;
- }
-
- stream_.mode = UNINITIALIZED;
- stream_.state = STREAM_CLOSED;
-}
-
-bool stopThreadCalled = false;
-
-void RtApiAsio :: startStream()
-{
- verifyStream();
- if ( stream_.state == STREAM_RUNNING ) {
- errorText_ = "RtApiAsio::startStream(): the stream is already running!";
- error( RtAudioError::WARNING );
- return;
- }
-
- 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;
- }
-
- handle->drainCounter = 0;
- handle->internalDrain = false;
- ResetEvent( handle->condition );
- stream_.state = STREAM_RUNNING;
- asioXRun = false;
-
- unlock:
- stopThreadCalled = false;
-
- if ( result == ASE_OK ) return;
- error( RtAudioError::SYSTEM_ERROR );
-}
-
-void RtApiAsio :: stopStream()
-{
- verifyStream();
- if ( stream_.state == STREAM_STOPPED ) {
- errorText_ = "RtApiAsio::stopStream(): the stream is already stopped!";
- error( RtAudioError::WARNING );
- return;
- }
-
- AsioHandle *handle = (AsioHandle *) stream_.apiHandle;
- if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
- if ( handle->drainCounter == 0 ) {
- handle->drainCounter = 2;
- WaitForSingleObject( handle->condition, INFINITE ); // block until signaled
- }
- }
-
- stream_.state = STREAM_STOPPED;
-
- ASIOError result = ASIOStop();
- if ( result != ASE_OK ) {
- errorStream_ << "RtApiAsio::stopStream: error (" << getAsioErrorString( result ) << ") stopping device.";
- errorText_ = errorStream_.str();
- }
-
- if ( result == ASE_OK ) return;
- error( RtAudioError::SYSTEM_ERROR );
-}
-
-void RtApiAsio :: abortStream()
-{
- verifyStream();
- if ( stream_.state == STREAM_STOPPED ) {
- errorText_ = "RtApiAsio::abortStream(): the stream is already stopped!";
- error( RtAudioError::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 = 2;
- stopStream();
-}
-
-// 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 ASIOStop()
-// function will return.
-static unsigned __stdcall asioStopStream( void *ptr )
-{
- CallbackInfo *info = (CallbackInfo *) ptr;
- RtApiAsio *object = (RtApiAsio *) info->object;
-
- object->stopStream();
- _endthreadex( 0 );
- return 0;
-}
-
-bool RtApiAsio :: callbackEvent( long bufferIndex )
-{
- if ( stream_.state == STREAM_STOPPED || stream_.state == STREAM_STOPPING ) return SUCCESS;
- if ( stream_.state == STREAM_CLOSED ) {
- errorText_ = "RtApiAsio::callbackEvent(): the stream is closed ... this shouldn't happen!";
- error( RtAudioError::WARNING );
- return FAILURE;
- }
-
- CallbackInfo *info = (CallbackInfo *) &stream_.callbackInfo;
- AsioHandle *handle = (AsioHandle *) stream_.apiHandle;
-
- // Check if we were draining the stream and signal if finished.
- if ( handle->drainCounter > 3 ) {
-
- stream_.state = STREAM_STOPPING;
- if ( handle->internalDrain == false )
- SetEvent( handle->condition );
- else { // spawn a thread to stop the stream
- unsigned threadId;
- stream_.callbackInfo.thread = _beginthreadex( NULL, 0, &asioStopStream,
- &stream_.callbackInfo, 0, &threadId );
- }
- return SUCCESS;
- }
-
- // 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;
- }
- int cbReturnValue = callback( stream_.userBuffer[0], stream_.userBuffer[1],
- stream_.bufferSize, streamTime, status, info->userData );
- if ( cbReturnValue == 2 ) {
- stream_.state = STREAM_STOPPING;
- handle->drainCounter = 2;
- unsigned threadId;
- stream_.callbackInfo.thread = _beginthreadex( NULL, 0, &asioStopStream,
- &stream_.callbackInfo, 0, &threadId );
- return SUCCESS;
- }
- else if ( cbReturnValue == 1 ) {
- 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 ( 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 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 {
-
- if ( stream_.doByteSwap[0] )
- byteSwapBuffer( stream_.userBuffer[0],
- stream_.bufferSize * stream_.nUserChannels[0],
- stream_.userFormat );
-
- 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 );
- }
-
- }
- }
-
- // Don't bother draining input
- if ( handle->drainCounter ) {
- handle->drainCounter++;
- goto unlock;
- }
-
- 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] );
-
- }
- 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();
-
- RtApi::tickStreamTime();
- return SUCCESS;
-}
-
-static 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 ( RtAudioError &exception ) {
- std::cerr << "\nRtApiAsio: sampleRateChanged() error (" << exception.getMessage() << ")!\n" << std::endl;
- return;
- }
-
- std::cerr << "\nRtApiAsio: driver reports sample rate changed to " << sRate << " ... stream stopped!!!\n" << std::endl;
-}
-
-static 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;
- 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 )
-{
- struct Messages
- {
- ASIOError value;
- const char*message;
- };
-
- static const 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;
-
- return "Unknown error.";
-}
-
-//******************** End of __WINDOWS_ASIO__ *********************//
-#endif
-
-
-#if defined(__WINDOWS_WASAPI__) // Windows WASAPI API
-
-// Authored by Marcus Tomlinson <themarcustomlinson@gmail.com>, April 2014
-// - Introduces support for the Windows WASAPI API
-// - Aims to deliver bit streams to and from hardware at the lowest possible latency, via the absolute minimum buffer sizes required
-// - Provides flexible stream configuration to an otherwise strict and inflexible WASAPI interface
-// - Includes automatic internal conversion of sample rate and buffer size between hardware and the user
-
-#ifndef INITGUID
- #define INITGUID
-#endif
-#include <audioclient.h>
-#include <avrt.h>
-#include <mmdeviceapi.h>
-#include <functiondiscoverykeys_devpkey.h>
-
-//=============================================================================
-
-#define SAFE_RELEASE( objectPtr )\
-if ( objectPtr )\
-{\
- objectPtr->Release();\
- objectPtr = NULL;\
-}
-
-typedef HANDLE ( __stdcall *TAvSetMmThreadCharacteristicsPtr )( LPCWSTR TaskName, LPDWORD TaskIndex );
-
-//-----------------------------------------------------------------------------
-
-// WASAPI dictates stream sample rate, format, channel count, and in some cases, buffer size.
-// Therefore we must perform all necessary conversions to user buffers in order to satisfy these
-// requirements. WasapiBuffer ring buffers are used between HwIn->UserIn and UserOut->HwOut to
-// provide intermediate storage for read / write synchronization.
-class WasapiBuffer
-{
-public:
- WasapiBuffer()
- : buffer_( NULL ),
- bufferSize_( 0 ),
- inIndex_( 0 ),
- outIndex_( 0 ) {}
-
- ~WasapiBuffer() {
- free( buffer_ );
- }
-
- // sets the length of the internal ring buffer
- void setBufferSize( unsigned int bufferSize, unsigned int formatBytes ) {
- free( buffer_ );
-
- buffer_ = ( char* ) calloc( bufferSize, formatBytes );
-
- bufferSize_ = bufferSize;
- inIndex_ = 0;
- outIndex_ = 0;
- }
-
- // attempt to push a buffer into the ring buffer at the current "in" index
- bool pushBuffer( char* buffer, unsigned int bufferSize, RtAudioFormat format )
- {
- if ( !buffer || // incoming buffer is NULL
- bufferSize == 0 || // incoming buffer has no data
- bufferSize > bufferSize_ ) // incoming buffer too large
- {
- return false;
- }
-
- unsigned int relOutIndex = outIndex_;
- unsigned int inIndexEnd = inIndex_ + bufferSize;
- if ( relOutIndex < inIndex_ && inIndexEnd >= bufferSize_ ) {
- relOutIndex += bufferSize_;
- }
-
- // "in" index can end on the "out" index but cannot begin at it
- if ( inIndex_ <= relOutIndex && inIndexEnd > relOutIndex ) {
- return false; // not enough space between "in" index and "out" index
- }
-
- // copy buffer from external to internal
- int fromZeroSize = inIndex_ + bufferSize - bufferSize_;
- fromZeroSize = fromZeroSize < 0 ? 0 : fromZeroSize;
- int fromInSize = bufferSize - fromZeroSize;
-
- switch( format )
- {
- case RTAUDIO_SINT8:
- memcpy( &( ( char* ) buffer_ )[inIndex_], buffer, fromInSize * sizeof( char ) );
- memcpy( buffer_, &( ( char* ) buffer )[fromInSize], fromZeroSize * sizeof( char ) );
- break;
- case RTAUDIO_SINT16:
- memcpy( &( ( short* ) buffer_ )[inIndex_], buffer, fromInSize * sizeof( short ) );
- memcpy( buffer_, &( ( short* ) buffer )[fromInSize], fromZeroSize * sizeof( short ) );
- break;
- case RTAUDIO_SINT24:
- memcpy( &( ( S24* ) buffer_ )[inIndex_], buffer, fromInSize * sizeof( S24 ) );
- memcpy( buffer_, &( ( S24* ) buffer )[fromInSize], fromZeroSize * sizeof( S24 ) );
- break;
- case RTAUDIO_SINT32:
- memcpy( &( ( int* ) buffer_ )[inIndex_], buffer, fromInSize * sizeof( int ) );
- memcpy( buffer_, &( ( int* ) buffer )[fromInSize], fromZeroSize * sizeof( int ) );
- break;
- case RTAUDIO_FLOAT32:
- memcpy( &( ( float* ) buffer_ )[inIndex_], buffer, fromInSize * sizeof( float ) );
- memcpy( buffer_, &( ( float* ) buffer )[fromInSize], fromZeroSize * sizeof( float ) );
- break;
- case RTAUDIO_FLOAT64:
- memcpy( &( ( double* ) buffer_ )[inIndex_], buffer, fromInSize * sizeof( double ) );
- memcpy( buffer_, &( ( double* ) buffer )[fromInSize], fromZeroSize * sizeof( double ) );
- break;
- }
-
- // update "in" index
- inIndex_ += bufferSize;
- inIndex_ %= bufferSize_;
-
- return true;
- }
-
- // attempt to pull a buffer from the ring buffer from the current "out" index
- bool pullBuffer( char* buffer, unsigned int bufferSize, RtAudioFormat format )
- {
- if ( !buffer || // incoming buffer is NULL
- bufferSize == 0 || // incoming buffer has no data
- bufferSize > bufferSize_ ) // incoming buffer too large
- {
- return false;
- }
-
- unsigned int relInIndex = inIndex_;
- unsigned int outIndexEnd = outIndex_ + bufferSize;
- if ( relInIndex < outIndex_ && outIndexEnd >= bufferSize_ ) {
- relInIndex += bufferSize_;
- }
-
- // "out" index can begin at and end on the "in" index
- if ( outIndex_ < relInIndex && outIndexEnd > relInIndex ) {
- return false; // not enough space between "out" index and "in" index
- }
-
- // copy buffer from internal to external
- int fromZeroSize = outIndex_ + bufferSize - bufferSize_;
- fromZeroSize = fromZeroSize < 0 ? 0 : fromZeroSize;
- int fromOutSize = bufferSize - fromZeroSize;
-
- switch( format )
- {
- case RTAUDIO_SINT8:
- memcpy( buffer, &( ( char* ) buffer_ )[outIndex_], fromOutSize * sizeof( char ) );
- memcpy( &( ( char* ) buffer )[fromOutSize], buffer_, fromZeroSize * sizeof( char ) );
- break;
- case RTAUDIO_SINT16:
- memcpy( buffer, &( ( short* ) buffer_ )[outIndex_], fromOutSize * sizeof( short ) );
- memcpy( &( ( short* ) buffer )[fromOutSize], buffer_, fromZeroSize * sizeof( short ) );
- break;
- case RTAUDIO_SINT24:
- memcpy( buffer, &( ( S24* ) buffer_ )[outIndex_], fromOutSize * sizeof( S24 ) );
- memcpy( &( ( S24* ) buffer )[fromOutSize], buffer_, fromZeroSize * sizeof( S24 ) );
- break;
- case RTAUDIO_SINT32:
- memcpy( buffer, &( ( int* ) buffer_ )[outIndex_], fromOutSize * sizeof( int ) );
- memcpy( &( ( int* ) buffer )[fromOutSize], buffer_, fromZeroSize * sizeof( int ) );
- break;
- case RTAUDIO_FLOAT32:
- memcpy( buffer, &( ( float* ) buffer_ )[outIndex_], fromOutSize * sizeof( float ) );
- memcpy( &( ( float* ) buffer )[fromOutSize], buffer_, fromZeroSize * sizeof( float ) );
- break;
- case RTAUDIO_FLOAT64:
- memcpy( buffer, &( ( double* ) buffer_ )[outIndex_], fromOutSize * sizeof( double ) );
- memcpy( &( ( double* ) buffer )[fromOutSize], buffer_, fromZeroSize * sizeof( double ) );
- break;
- }
-
- // update "out" index
- outIndex_ += bufferSize;
- outIndex_ %= bufferSize_;
-
- return true;
- }
-
-private:
- char* buffer_;
- unsigned int bufferSize_;
- unsigned int inIndex_;
- unsigned int outIndex_;
-};
-
-//-----------------------------------------------------------------------------
-
-// In order to satisfy WASAPI's buffer requirements, we need a means of converting sample rate
-// between HW and the user. The convertBufferWasapi function is used to perform this conversion
-// between HwIn->UserIn and UserOut->HwOut during the stream callback loop.
-// This sample rate converter favors speed over quality, and works best with conversions between
-// one rate and its multiple.
-void convertBufferWasapi( char* outBuffer,
- const char* inBuffer,
- const unsigned int& channelCount,
- const unsigned int& inSampleRate,
- const unsigned int& outSampleRate,
- const unsigned int& inSampleCount,
- unsigned int& outSampleCount,
- const RtAudioFormat& format )
-{
- // calculate the new outSampleCount and relative sampleStep
- float sampleRatio = ( float ) outSampleRate / inSampleRate;
- float sampleStep = 1.0f / sampleRatio;
- float inSampleFraction = 0.0f;
-
- outSampleCount = ( unsigned int ) roundf( inSampleCount * sampleRatio );
-
- // frame-by-frame, copy each relative input sample into it's corresponding output sample
- for ( unsigned int outSample = 0; outSample < outSampleCount; outSample++ )
- {
- unsigned int inSample = ( unsigned int ) inSampleFraction;
-
- switch ( format )
- {
- case RTAUDIO_SINT8:
- memcpy( &( ( char* ) outBuffer )[ outSample * channelCount ], &( ( char* ) inBuffer )[ inSample * channelCount ], channelCount * sizeof( char ) );
- break;
- case RTAUDIO_SINT16:
- memcpy( &( ( short* ) outBuffer )[ outSample * channelCount ], &( ( short* ) inBuffer )[ inSample * channelCount ], channelCount * sizeof( short ) );
- break;
- case RTAUDIO_SINT24:
- memcpy( &( ( S24* ) outBuffer )[ outSample * channelCount ], &( ( S24* ) inBuffer )[ inSample * channelCount ], channelCount * sizeof( S24 ) );
- break;
- case RTAUDIO_SINT32:
- memcpy( &( ( int* ) outBuffer )[ outSample * channelCount ], &( ( int* ) inBuffer )[ inSample * channelCount ], channelCount * sizeof( int ) );
- break;
- case RTAUDIO_FLOAT32:
- memcpy( &( ( float* ) outBuffer )[ outSample * channelCount ], &( ( float* ) inBuffer )[ inSample * channelCount ], channelCount * sizeof( float ) );
- break;
- case RTAUDIO_FLOAT64:
- memcpy( &( ( double* ) outBuffer )[ outSample * channelCount ], &( ( double* ) inBuffer )[ inSample * channelCount ], channelCount * sizeof( double ) );
- break;
- }
-
- // jump to next in sample
- inSampleFraction += sampleStep;
- }
-}
-
-//-----------------------------------------------------------------------------
-
-// A structure to hold various information related to the WASAPI implementation.
-struct WasapiHandle
-{
- IAudioClient* captureAudioClient;
- IAudioClient* renderAudioClient;
- IAudioCaptureClient* captureClient;
- IAudioRenderClient* renderClient;
- HANDLE captureEvent;
- HANDLE renderEvent;
-
- WasapiHandle()
- : captureAudioClient( NULL ),
- renderAudioClient( NULL ),
- captureClient( NULL ),
- renderClient( NULL ),
- captureEvent( NULL ),
- renderEvent( NULL ) {}
-};
-
-//=============================================================================
-
-RtApiWasapi::RtApiWasapi()
- : coInitialized_( false ), deviceEnumerator_( NULL )
-{
- // WASAPI can run either apartment or multi-threaded
- HRESULT hr = CoInitialize( NULL );
- if ( !FAILED( hr ) )
- coInitialized_ = true;
-
- // Instantiate device enumerator
- hr = CoCreateInstance( __uuidof( MMDeviceEnumerator ), NULL,
- CLSCTX_ALL, __uuidof( IMMDeviceEnumerator ),
- ( void** ) &deviceEnumerator_ );
-
- if ( FAILED( hr ) ) {
- errorText_ = "RtApiWasapi::RtApiWasapi: Unable to instantiate device enumerator";
- error( RtAudioError::DRIVER_ERROR );
- }
-}
-
-//-----------------------------------------------------------------------------
-
-RtApiWasapi::~RtApiWasapi()
-{
- if ( stream_.state != STREAM_CLOSED )
- closeStream();
-
- SAFE_RELEASE( deviceEnumerator_ );
-
- // If this object previously called CoInitialize()
- if ( coInitialized_ )
- CoUninitialize();
-}
-
-//=============================================================================
-
-unsigned int RtApiWasapi::getDeviceCount( void )
-{
- unsigned int captureDeviceCount = 0;
- unsigned int renderDeviceCount = 0;
-
- IMMDeviceCollection* captureDevices = NULL;
- IMMDeviceCollection* renderDevices = NULL;
-
- // Count capture devices
- errorText_.clear();
- HRESULT hr = deviceEnumerator_->EnumAudioEndpoints( eCapture, DEVICE_STATE_ACTIVE, &captureDevices );
- if ( FAILED( hr ) ) {
- errorText_ = "RtApiWasapi::getDeviceCount: Unable to retrieve capture device collection.";
- goto Exit;
- }
-
- hr = captureDevices->GetCount( &captureDeviceCount );
- if ( FAILED( hr ) ) {
- errorText_ = "RtApiWasapi::getDeviceCount: Unable to retrieve capture device count.";
- goto Exit;
- }
-
- // Count render devices
- hr = deviceEnumerator_->EnumAudioEndpoints( eRender, DEVICE_STATE_ACTIVE, &renderDevices );
- if ( FAILED( hr ) ) {
- errorText_ = "RtApiWasapi::getDeviceCount: Unable to retrieve render device collection.";
- goto Exit;
- }
-
- hr = renderDevices->GetCount( &renderDeviceCount );
- if ( FAILED( hr ) ) {
- errorText_ = "RtApiWasapi::getDeviceCount: Unable to retrieve render device count.";
- goto Exit;
- }
-
-Exit:
- // release all references
- SAFE_RELEASE( captureDevices );
- SAFE_RELEASE( renderDevices );
-
- if ( errorText_.empty() )
- return captureDeviceCount + renderDeviceCount;
-
- error( RtAudioError::DRIVER_ERROR );
- return 0;
-}
-
-//-----------------------------------------------------------------------------
-
-RtAudio::DeviceInfo RtApiWasapi::getDeviceInfo( unsigned int device )
-{
- RtAudio::DeviceInfo info;
- unsigned int captureDeviceCount = 0;
- unsigned int renderDeviceCount = 0;
- std::string defaultDeviceName;
- bool isCaptureDevice = false;
-
- PROPVARIANT deviceNameProp;
- PROPVARIANT defaultDeviceNameProp;
-
- IMMDeviceCollection* captureDevices = NULL;
- IMMDeviceCollection* renderDevices = NULL;
- IMMDevice* devicePtr = NULL;
- IMMDevice* defaultDevicePtr = NULL;
- IAudioClient* audioClient = NULL;
- IPropertyStore* devicePropStore = NULL;
- IPropertyStore* defaultDevicePropStore = NULL;
-
- WAVEFORMATEX* deviceFormat = NULL;
- WAVEFORMATEX* closestMatchFormat = NULL;
-
- // probed
- info.probed = false;
-
- // Count capture devices
- errorText_.clear();
- RtAudioError::Type errorType = RtAudioError::DRIVER_ERROR;
- HRESULT hr = deviceEnumerator_->EnumAudioEndpoints( eCapture, DEVICE_STATE_ACTIVE, &captureDevices );
- if ( FAILED( hr ) ) {
- errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve capture device collection.";
- goto Exit;
- }
-
- hr = captureDevices->GetCount( &captureDeviceCount );
- if ( FAILED( hr ) ) {
- errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve capture device count.";
- goto Exit;
- }
-
- // Count render devices
- hr = deviceEnumerator_->EnumAudioEndpoints( eRender, DEVICE_STATE_ACTIVE, &renderDevices );
- if ( FAILED( hr ) ) {
- errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve render device collection.";
- goto Exit;
- }
-
- hr = renderDevices->GetCount( &renderDeviceCount );
- if ( FAILED( hr ) ) {
- errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve render device count.";
- goto Exit;
- }
-
- // validate device index
- if ( device >= captureDeviceCount + renderDeviceCount ) {
- errorText_ = "RtApiWasapi::getDeviceInfo: Invalid device index.";
- errorType = RtAudioError::INVALID_USE;
- goto Exit;
- }
-
- // determine whether index falls within capture or render devices
- if ( device >= renderDeviceCount ) {
- hr = captureDevices->Item( device - renderDeviceCount, &devicePtr );
- if ( FAILED( hr ) ) {
- errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve capture device handle.";
- goto Exit;
- }
- isCaptureDevice = true;
- }
- else {
- hr = renderDevices->Item( device, &devicePtr );
- if ( FAILED( hr ) ) {
- errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve render device handle.";
- goto Exit;
- }
- isCaptureDevice = false;
- }
-
- // get default device name
- if ( isCaptureDevice ) {
- hr = deviceEnumerator_->GetDefaultAudioEndpoint( eCapture, eConsole, &defaultDevicePtr );
- if ( FAILED( hr ) ) {
- errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve default capture device handle.";
- goto Exit;
- }
- }
- else {
- hr = deviceEnumerator_->GetDefaultAudioEndpoint( eRender, eConsole, &defaultDevicePtr );
- if ( FAILED( hr ) ) {
- errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve default render device handle.";
- goto Exit;
- }
- }
-
- hr = defaultDevicePtr->OpenPropertyStore( STGM_READ, &defaultDevicePropStore );
- if ( FAILED( hr ) ) {
- errorText_ = "RtApiWasapi::getDeviceInfo: Unable to open default device property store.";
- goto Exit;
- }
- PropVariantInit( &defaultDeviceNameProp );
-
- hr = defaultDevicePropStore->GetValue( PKEY_Device_FriendlyName, &defaultDeviceNameProp );
- if ( FAILED( hr ) ) {
- errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve default device property: PKEY_Device_FriendlyName.";
- goto Exit;
- }
-
- defaultDeviceName = convertCharPointerToStdString(defaultDeviceNameProp.pwszVal);
-
- // name
- hr = devicePtr->OpenPropertyStore( STGM_READ, &devicePropStore );
- if ( FAILED( hr ) ) {
- errorText_ = "RtApiWasapi::getDeviceInfo: Unable to open device property store.";
- goto Exit;
- }
-
- PropVariantInit( &deviceNameProp );
-
- hr = devicePropStore->GetValue( PKEY_Device_FriendlyName, &deviceNameProp );
- if ( FAILED( hr ) ) {
- errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve device property: PKEY_Device_FriendlyName.";
- goto Exit;
- }
-
- info.name =convertCharPointerToStdString(deviceNameProp.pwszVal);
-
- // is default
- if ( isCaptureDevice ) {
- info.isDefaultInput = info.name == defaultDeviceName;
- info.isDefaultOutput = false;
- }
- else {
- info.isDefaultInput = false;
- info.isDefaultOutput = info.name == defaultDeviceName;
- }
-
- // channel count
- hr = devicePtr->Activate( __uuidof( IAudioClient ), CLSCTX_ALL, NULL, ( void** ) &audioClient );
- if ( FAILED( hr ) ) {
- errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve device audio client.";
- goto Exit;
- }
-
- hr = audioClient->GetMixFormat( &deviceFormat );
- if ( FAILED( hr ) ) {
- errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve device mix format.";
- goto Exit;
- }
-
- if ( isCaptureDevice ) {
- info.inputChannels = deviceFormat->nChannels;
- info.outputChannels = 0;
- info.duplexChannels = 0;
- }
- else {
- info.inputChannels = 0;
- info.outputChannels = deviceFormat->nChannels;
- info.duplexChannels = 0;
- }
-
- // sample rates
- info.sampleRates.clear();
-
- // allow support for all sample rates as we have a built-in sample rate converter
- for ( unsigned int i = 0; i < MAX_SAMPLE_RATES; i++ ) {
- info.sampleRates.push_back( SAMPLE_RATES[i] );
- }
- info.preferredSampleRate = deviceFormat->nSamplesPerSec;
-
- // native format
- info.nativeFormats = 0;
-
- if ( deviceFormat->wFormatTag == WAVE_FORMAT_IEEE_FLOAT ||
- ( deviceFormat->wFormatTag == WAVE_FORMAT_EXTENSIBLE &&
- ( ( WAVEFORMATEXTENSIBLE* ) deviceFormat )->SubFormat == KSDATAFORMAT_SUBTYPE_IEEE_FLOAT ) )
- {
- if ( deviceFormat->wBitsPerSample == 32 ) {
- info.nativeFormats |= RTAUDIO_FLOAT32;
- }
- else if ( deviceFormat->wBitsPerSample == 64 ) {
- info.nativeFormats |= RTAUDIO_FLOAT64;
- }
- }
- else if ( deviceFormat->wFormatTag == WAVE_FORMAT_PCM ||
- ( deviceFormat->wFormatTag == WAVE_FORMAT_EXTENSIBLE &&
- ( ( WAVEFORMATEXTENSIBLE* ) deviceFormat )->SubFormat == KSDATAFORMAT_SUBTYPE_PCM ) )
- {
- if ( deviceFormat->wBitsPerSample == 8 ) {
- info.nativeFormats |= RTAUDIO_SINT8;
- }
- else if ( deviceFormat->wBitsPerSample == 16 ) {
- info.nativeFormats |= RTAUDIO_SINT16;
- }
- else if ( deviceFormat->wBitsPerSample == 24 ) {
- info.nativeFormats |= RTAUDIO_SINT24;
- }
- else if ( deviceFormat->wBitsPerSample == 32 ) {
- info.nativeFormats |= RTAUDIO_SINT32;
- }
- }
-
- // probed
- info.probed = true;
-
-Exit:
- // release all references
- PropVariantClear( &deviceNameProp );
- PropVariantClear( &defaultDeviceNameProp );
-
- SAFE_RELEASE( captureDevices );
- SAFE_RELEASE( renderDevices );
- SAFE_RELEASE( devicePtr );
- SAFE_RELEASE( defaultDevicePtr );
- SAFE_RELEASE( audioClient );
- SAFE_RELEASE( devicePropStore );
- SAFE_RELEASE( defaultDevicePropStore );
-
- CoTaskMemFree( deviceFormat );
- CoTaskMemFree( closestMatchFormat );
-
- if ( !errorText_.empty() )
- error( errorType );
- return info;
-}
-
-//-----------------------------------------------------------------------------
-
-unsigned int RtApiWasapi::getDefaultOutputDevice( void )
-{
- for ( unsigned int i = 0; i < getDeviceCount(); i++ ) {
- if ( getDeviceInfo( i ).isDefaultOutput ) {
- return i;
- }
- }
-
- return 0;
-}
-
-//-----------------------------------------------------------------------------
-
-unsigned int RtApiWasapi::getDefaultInputDevice( void )
-{
- for ( unsigned int i = 0; i < getDeviceCount(); i++ ) {
- if ( getDeviceInfo( i ).isDefaultInput ) {
- return i;
- }
- }
-
- return 0;
-}
-
-//-----------------------------------------------------------------------------
-
-void RtApiWasapi::closeStream( void )
-{
- if ( stream_.state == STREAM_CLOSED ) {
- errorText_ = "RtApiWasapi::closeStream: No open stream to close.";
- error( RtAudioError::WARNING );
- return;
- }
-
- if ( stream_.state != STREAM_STOPPED )
- stopStream();
-
- // clean up stream memory
- SAFE_RELEASE( ( ( WasapiHandle* ) stream_.apiHandle )->captureAudioClient )
- SAFE_RELEASE( ( ( WasapiHandle* ) stream_.apiHandle )->renderAudioClient )
-
- SAFE_RELEASE( ( ( WasapiHandle* ) stream_.apiHandle )->captureClient )
- SAFE_RELEASE( ( ( WasapiHandle* ) stream_.apiHandle )->renderClient )
-
- if ( ( ( WasapiHandle* ) stream_.apiHandle )->captureEvent )
- CloseHandle( ( ( WasapiHandle* ) stream_.apiHandle )->captureEvent );
-
- if ( ( ( WasapiHandle* ) stream_.apiHandle )->renderEvent )
- CloseHandle( ( ( WasapiHandle* ) stream_.apiHandle )->renderEvent );
-
- delete ( WasapiHandle* ) stream_.apiHandle;
- stream_.apiHandle = NULL;
-
- 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;
- }
-
- // update stream state
- stream_.state = STREAM_CLOSED;
-}
-
-//-----------------------------------------------------------------------------
-
-void RtApiWasapi::startStream( void )
-{
- verifyStream();
-
- if ( stream_.state == STREAM_RUNNING ) {
- errorText_ = "RtApiWasapi::startStream: The stream is already running.";
- error( RtAudioError::WARNING );
- return;
- }
-
- // update stream state
- stream_.state = STREAM_RUNNING;
-
- // create WASAPI stream thread
- stream_.callbackInfo.thread = ( ThreadHandle ) CreateThread( NULL, 0, runWasapiThread, this, CREATE_SUSPENDED, NULL );
-
- if ( !stream_.callbackInfo.thread ) {
- errorText_ = "RtApiWasapi::startStream: Unable to instantiate callback thread.";
- error( RtAudioError::THREAD_ERROR );
- }
- else {
- SetThreadPriority( ( void* ) stream_.callbackInfo.thread, stream_.callbackInfo.priority );
- ResumeThread( ( void* ) stream_.callbackInfo.thread );
- }
-}
-
-//-----------------------------------------------------------------------------
-
-void RtApiWasapi::stopStream( void )
-{
- verifyStream();
-
- if ( stream_.state == STREAM_STOPPED ) {
- errorText_ = "RtApiWasapi::stopStream: The stream is already stopped.";
- error( RtAudioError::WARNING );
- return;
- }
-
- // inform stream thread by setting stream state to STREAM_STOPPING
- stream_.state = STREAM_STOPPING;
-
- // wait until stream thread is stopped
- while( stream_.state != STREAM_STOPPED ) {
- Sleep( 1 );
- }
-
- // Wait for the last buffer to play before stopping.
- Sleep( 1000 * stream_.bufferSize / stream_.sampleRate );
-
- // stop capture client if applicable
- if ( ( ( WasapiHandle* ) stream_.apiHandle )->captureAudioClient ) {
- HRESULT hr = ( ( WasapiHandle* ) stream_.apiHandle )->captureAudioClient->Stop();
- if ( FAILED( hr ) ) {
- errorText_ = "RtApiWasapi::stopStream: Unable to stop capture stream.";
- error( RtAudioError::DRIVER_ERROR );
- return;
- }
- }
-
- // stop render client if applicable
- if ( ( ( WasapiHandle* ) stream_.apiHandle )->renderAudioClient ) {
- HRESULT hr = ( ( WasapiHandle* ) stream_.apiHandle )->renderAudioClient->Stop();
- if ( FAILED( hr ) ) {
- errorText_ = "RtApiWasapi::stopStream: Unable to stop render stream.";
- error( RtAudioError::DRIVER_ERROR );
- return;
- }
- }
-
- // close thread handle
- if ( stream_.callbackInfo.thread && !CloseHandle( ( void* ) stream_.callbackInfo.thread ) ) {
- errorText_ = "RtApiWasapi::stopStream: Unable to close callback thread.";
- error( RtAudioError::THREAD_ERROR );
- return;
- }
-
- stream_.callbackInfo.thread = (ThreadHandle) NULL;
-}
-
-//-----------------------------------------------------------------------------
-
-void RtApiWasapi::abortStream( void )
-{
- verifyStream();
-
- if ( stream_.state == STREAM_STOPPED ) {
- errorText_ = "RtApiWasapi::abortStream: The stream is already stopped.";
- error( RtAudioError::WARNING );
- return;
- }
-
- // inform stream thread by setting stream state to STREAM_STOPPING
- stream_.state = STREAM_STOPPING;
-
- // wait until stream thread is stopped
- while ( stream_.state != STREAM_STOPPED ) {
- Sleep( 1 );
- }
-
- // stop capture client if applicable
- if ( ( ( WasapiHandle* ) stream_.apiHandle )->captureAudioClient ) {
- HRESULT hr = ( ( WasapiHandle* ) stream_.apiHandle )->captureAudioClient->Stop();
- if ( FAILED( hr ) ) {
- errorText_ = "RtApiWasapi::abortStream: Unable to stop capture stream.";
- error( RtAudioError::DRIVER_ERROR );
- return;
- }
- }
-
- // stop render client if applicable
- if ( ( ( WasapiHandle* ) stream_.apiHandle )->renderAudioClient ) {
- HRESULT hr = ( ( WasapiHandle* ) stream_.apiHandle )->renderAudioClient->Stop();
- if ( FAILED( hr ) ) {
- errorText_ = "RtApiWasapi::abortStream: Unable to stop render stream.";
- error( RtAudioError::DRIVER_ERROR );
- return;
- }
- }
-
- // close thread handle
- if ( stream_.callbackInfo.thread && !CloseHandle( ( void* ) stream_.callbackInfo.thread ) ) {
- errorText_ = "RtApiWasapi::abortStream: Unable to close callback thread.";
- error( RtAudioError::THREAD_ERROR );
- return;
- }
-
- stream_.callbackInfo.thread = (ThreadHandle) NULL;
-}
-
-//-----------------------------------------------------------------------------
-
-bool RtApiWasapi::probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,
- unsigned int firstChannel, unsigned int sampleRate,
- RtAudioFormat format, unsigned int* bufferSize,
- RtAudio::StreamOptions* options )
-{
- bool methodResult = FAILURE;
- unsigned int captureDeviceCount = 0;
- unsigned int renderDeviceCount = 0;
-
- IMMDeviceCollection* captureDevices = NULL;
- IMMDeviceCollection* renderDevices = NULL;
- IMMDevice* devicePtr = NULL;
- WAVEFORMATEX* deviceFormat = NULL;
- unsigned int bufferBytes;
- stream_.state = STREAM_STOPPED;
-
- // create API Handle if not already created
- if ( !stream_.apiHandle )
- stream_.apiHandle = ( void* ) new WasapiHandle();
-
- // Count capture devices
- errorText_.clear();
- RtAudioError::Type errorType = RtAudioError::DRIVER_ERROR;
- HRESULT hr = deviceEnumerator_->EnumAudioEndpoints( eCapture, DEVICE_STATE_ACTIVE, &captureDevices );
- if ( FAILED( hr ) ) {
- errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve capture device collection.";
- goto Exit;
- }
-
- hr = captureDevices->GetCount( &captureDeviceCount );
- if ( FAILED( hr ) ) {
- errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve capture device count.";
- goto Exit;
- }
-
- // Count render devices
- hr = deviceEnumerator_->EnumAudioEndpoints( eRender, DEVICE_STATE_ACTIVE, &renderDevices );
- if ( FAILED( hr ) ) {
- errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve render device collection.";
- goto Exit;
- }
-
- hr = renderDevices->GetCount( &renderDeviceCount );
- if ( FAILED( hr ) ) {
- errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve render device count.";
- goto Exit;
- }
-
- // validate device index
- if ( device >= captureDeviceCount + renderDeviceCount ) {
- errorType = RtAudioError::INVALID_USE;
- errorText_ = "RtApiWasapi::probeDeviceOpen: Invalid device index.";
- goto Exit;
- }
-
- // determine whether index falls within capture or render devices
- if ( device >= renderDeviceCount ) {
- if ( mode != INPUT ) {
- errorType = RtAudioError::INVALID_USE;
- errorText_ = "RtApiWasapi::probeDeviceOpen: Capture device selected as output device.";
- goto Exit;
- }
-
- // retrieve captureAudioClient from devicePtr
- IAudioClient*& captureAudioClient = ( ( WasapiHandle* ) stream_.apiHandle )->captureAudioClient;
-
- hr = captureDevices->Item( device - renderDeviceCount, &devicePtr );
- if ( FAILED( hr ) ) {
- errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve capture device handle.";
- goto Exit;
- }
-
- hr = devicePtr->Activate( __uuidof( IAudioClient ), CLSCTX_ALL,
- NULL, ( void** ) &captureAudioClient );
- if ( FAILED( hr ) ) {
- errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve device audio client.";
- goto Exit;
- }
-
- hr = captureAudioClient->GetMixFormat( &deviceFormat );
- if ( FAILED( hr ) ) {
- errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve device mix format.";
- goto Exit;
- }
-
- stream_.nDeviceChannels[mode] = deviceFormat->nChannels;
- captureAudioClient->GetStreamLatency( ( long long* ) &stream_.latency[mode] );
- }
- else {
- if ( mode != OUTPUT ) {
- errorType = RtAudioError::INVALID_USE;
- errorText_ = "RtApiWasapi::probeDeviceOpen: Render device selected as input device.";
- goto Exit;
- }
-
- // retrieve renderAudioClient from devicePtr
- IAudioClient*& renderAudioClient = ( ( WasapiHandle* ) stream_.apiHandle )->renderAudioClient;
-
- hr = renderDevices->Item( device, &devicePtr );
- if ( FAILED( hr ) ) {
- errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve render device handle.";
- goto Exit;
- }
-
- hr = devicePtr->Activate( __uuidof( IAudioClient ), CLSCTX_ALL,
- NULL, ( void** ) &renderAudioClient );
- if ( FAILED( hr ) ) {
- errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve device audio client.";
- goto Exit;
- }
-
- hr = renderAudioClient->GetMixFormat( &deviceFormat );
- if ( FAILED( hr ) ) {
- errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve device mix format.";
- goto Exit;
- }
-
- stream_.nDeviceChannels[mode] = deviceFormat->nChannels;
- renderAudioClient->GetStreamLatency( ( long long* ) &stream_.latency[mode] );
- }
-
- // fill stream data
- if ( ( stream_.mode == OUTPUT && mode == INPUT ) ||
- ( stream_.mode == INPUT && mode == OUTPUT ) ) {
- stream_.mode = DUPLEX;
- }
- else {
- stream_.mode = mode;
- }
-
- stream_.device[mode] = device;
- stream_.doByteSwap[mode] = false;
- stream_.sampleRate = sampleRate;
- stream_.bufferSize = *bufferSize;
- stream_.nBuffers = 1;
- stream_.nUserChannels[mode] = channels;
- stream_.channelOffset[mode] = firstChannel;
- stream_.userFormat = format;
- stream_.deviceFormat[mode] = getDeviceInfo( device ).nativeFormats;
-
- if ( options && options->flags & RTAUDIO_NONINTERLEAVED )
- stream_.userInterleaved = false;
- else
- stream_.userInterleaved = true;
- stream_.deviceInterleaved[mode] = true;
-
- // Set flags for buffer conversion.
- stream_.doConvertBuffer[mode] = false;
- if ( stream_.userFormat != stream_.deviceFormat[mode] ||
- stream_.nUserChannels != stream_.nDeviceChannels )
- stream_.doConvertBuffer[mode] = true;
- else if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] &&
- stream_.nUserChannels[mode] > 1 )
- stream_.doConvertBuffer[mode] = true;
-
- if ( stream_.doConvertBuffer[mode] )
- setConvertInfo( mode, 0 );
-
- // Allocate necessary internal buffers
- bufferBytes = stream_.nUserChannels[mode] * stream_.bufferSize * formatBytes( stream_.userFormat );
-
- stream_.userBuffer[mode] = ( char* ) calloc( bufferBytes, 1 );
- if ( !stream_.userBuffer[mode] ) {
- errorType = RtAudioError::MEMORY_ERROR;
- errorText_ = "RtApiWasapi::probeDeviceOpen: Error allocating user buffer memory.";
- goto Exit;
- }
-
- if ( options && options->flags & RTAUDIO_SCHEDULE_REALTIME )
- stream_.callbackInfo.priority = 15;
- else
- stream_.callbackInfo.priority = 0;
-
- ///! TODO: RTAUDIO_MINIMIZE_LATENCY // Provide stream buffers directly to callback
- ///! TODO: RTAUDIO_HOG_DEVICE // Exclusive mode
-
- methodResult = SUCCESS;
-
-Exit:
- //clean up
- SAFE_RELEASE( captureDevices );
- SAFE_RELEASE( renderDevices );
- SAFE_RELEASE( devicePtr );
- CoTaskMemFree( deviceFormat );
-
- // if method failed, close the stream
- if ( methodResult == FAILURE )
- closeStream();
-
- if ( !errorText_.empty() )
- error( errorType );
- return methodResult;
-}
-
-//=============================================================================
-
-DWORD WINAPI RtApiWasapi::runWasapiThread( void* wasapiPtr )
-{
- if ( wasapiPtr )
- ( ( RtApiWasapi* ) wasapiPtr )->wasapiThread();
-
- return 0;
-}
-
-DWORD WINAPI RtApiWasapi::stopWasapiThread( void* wasapiPtr )
-{
- if ( wasapiPtr )
- ( ( RtApiWasapi* ) wasapiPtr )->stopStream();
-
- return 0;
-}
-
-DWORD WINAPI RtApiWasapi::abortWasapiThread( void* wasapiPtr )
-{
- if ( wasapiPtr )
- ( ( RtApiWasapi* ) wasapiPtr )->abortStream();
-
- return 0;
-}
-
-//-----------------------------------------------------------------------------
-
-void RtApiWasapi::wasapiThread()
-{
- // as this is a new thread, we must CoInitialize it
- CoInitialize( NULL );
-
- HRESULT hr;
-
- IAudioClient* captureAudioClient = ( ( WasapiHandle* ) stream_.apiHandle )->captureAudioClient;
- IAudioClient* renderAudioClient = ( ( WasapiHandle* ) stream_.apiHandle )->renderAudioClient;
- IAudioCaptureClient* captureClient = ( ( WasapiHandle* ) stream_.apiHandle )->captureClient;
- IAudioRenderClient* renderClient = ( ( WasapiHandle* ) stream_.apiHandle )->renderClient;
- HANDLE captureEvent = ( ( WasapiHandle* ) stream_.apiHandle )->captureEvent;
- HANDLE renderEvent = ( ( WasapiHandle* ) stream_.apiHandle )->renderEvent;
-
- WAVEFORMATEX* captureFormat = NULL;
- WAVEFORMATEX* renderFormat = NULL;
- float captureSrRatio = 0.0f;
- float renderSrRatio = 0.0f;
- WasapiBuffer captureBuffer;
- WasapiBuffer renderBuffer;
-
- // declare local stream variables
- RtAudioCallback callback = ( RtAudioCallback ) stream_.callbackInfo.callback;
- BYTE* streamBuffer = NULL;
- unsigned long captureFlags = 0;
- unsigned int bufferFrameCount = 0;
- unsigned int numFramesPadding = 0;
- unsigned int convBufferSize = 0;
- bool callbackPushed = false;
- bool callbackPulled = false;
- bool callbackStopped = false;
- int callbackResult = 0;
-
- // convBuffer is used to store converted buffers between WASAPI and the user
- char* convBuffer = NULL;
- unsigned int convBuffSize = 0;
- unsigned int deviceBuffSize = 0;
-
- errorText_.clear();
- RtAudioError::Type errorType = RtAudioError::DRIVER_ERROR;
-
- // Attempt to assign "Pro Audio" characteristic to thread
- HMODULE AvrtDll = LoadLibrary( (LPCTSTR) "AVRT.dll" );
- if ( AvrtDll ) {
- DWORD taskIndex = 0;
- TAvSetMmThreadCharacteristicsPtr AvSetMmThreadCharacteristicsPtr = ( TAvSetMmThreadCharacteristicsPtr ) GetProcAddress( AvrtDll, "AvSetMmThreadCharacteristicsW" );
- AvSetMmThreadCharacteristicsPtr( L"Pro Audio", &taskIndex );
- FreeLibrary( AvrtDll );
- }
-
- // start capture stream if applicable
- if ( captureAudioClient ) {
- hr = captureAudioClient->GetMixFormat( &captureFormat );
- if ( FAILED( hr ) ) {
- errorText_ = "RtApiWasapi::wasapiThread: Unable to retrieve device mix format.";
- goto Exit;
- }
-
- captureSrRatio = ( ( float ) captureFormat->nSamplesPerSec / stream_.sampleRate );
-
- // initialize capture stream according to desire buffer size
- float desiredBufferSize = stream_.bufferSize * captureSrRatio;
- REFERENCE_TIME desiredBufferPeriod = ( REFERENCE_TIME ) ( ( float ) desiredBufferSize * 10000000 / captureFormat->nSamplesPerSec );
-
- if ( !captureClient ) {
- hr = captureAudioClient->Initialize( AUDCLNT_SHAREMODE_SHARED,
- AUDCLNT_STREAMFLAGS_EVENTCALLBACK,
- desiredBufferPeriod,
- desiredBufferPeriod,
- captureFormat,
- NULL );
- if ( FAILED( hr ) ) {
- errorText_ = "RtApiWasapi::wasapiThread: Unable to initialize capture audio client.";
- goto Exit;
- }
-
- hr = captureAudioClient->GetService( __uuidof( IAudioCaptureClient ),
- ( void** ) &captureClient );
- if ( FAILED( hr ) ) {
- errorText_ = "RtApiWasapi::wasapiThread: Unable to retrieve capture client handle.";
- goto Exit;
- }
-
- // configure captureEvent to trigger on every available capture buffer
- captureEvent = CreateEvent( NULL, FALSE, FALSE, NULL );
- if ( !captureEvent ) {
- errorType = RtAudioError::SYSTEM_ERROR;
- errorText_ = "RtApiWasapi::wasapiThread: Unable to create capture event.";
- goto Exit;
- }
-
- hr = captureAudioClient->SetEventHandle( captureEvent );
- if ( FAILED( hr ) ) {
- errorText_ = "RtApiWasapi::wasapiThread: Unable to set capture event handle.";
- goto Exit;
- }
-
- ( ( WasapiHandle* ) stream_.apiHandle )->captureClient = captureClient;
- ( ( WasapiHandle* ) stream_.apiHandle )->captureEvent = captureEvent;
- }
-
- unsigned int inBufferSize = 0;
- hr = captureAudioClient->GetBufferSize( &inBufferSize );
- if ( FAILED( hr ) ) {
- errorText_ = "RtApiWasapi::wasapiThread: Unable to get capture buffer size.";
- goto Exit;
- }
-
- // scale outBufferSize according to stream->user sample rate ratio
- unsigned int outBufferSize = ( unsigned int ) ( stream_.bufferSize * captureSrRatio ) * stream_.nDeviceChannels[INPUT];
- inBufferSize *= stream_.nDeviceChannels[INPUT];
-
- // set captureBuffer size
- captureBuffer.setBufferSize( inBufferSize + outBufferSize, formatBytes( stream_.deviceFormat[INPUT] ) );
-
- // reset the capture stream
- hr = captureAudioClient->Reset();
- if ( FAILED( hr ) ) {
- errorText_ = "RtApiWasapi::wasapiThread: Unable to reset capture stream.";
- goto Exit;
- }
-
- // start the capture stream
- hr = captureAudioClient->Start();
- if ( FAILED( hr ) ) {
- errorText_ = "RtApiWasapi::wasapiThread: Unable to start capture stream.";
- goto Exit;
- }
- }
-
- // start render stream if applicable
- if ( renderAudioClient ) {
- hr = renderAudioClient->GetMixFormat( &renderFormat );
- if ( FAILED( hr ) ) {
- errorText_ = "RtApiWasapi::wasapiThread: Unable to retrieve device mix format.";
- goto Exit;
- }
-
- renderSrRatio = ( ( float ) renderFormat->nSamplesPerSec / stream_.sampleRate );
-
- // initialize render stream according to desire buffer size
- float desiredBufferSize = stream_.bufferSize * renderSrRatio;
- REFERENCE_TIME desiredBufferPeriod = ( REFERENCE_TIME ) ( ( float ) desiredBufferSize * 10000000 / renderFormat->nSamplesPerSec );
-
- if ( !renderClient ) {
- hr = renderAudioClient->Initialize( AUDCLNT_SHAREMODE_SHARED,
- AUDCLNT_STREAMFLAGS_EVENTCALLBACK,
- desiredBufferPeriod,
- desiredBufferPeriod,
- renderFormat,
- NULL );
- if ( FAILED( hr ) ) {
- errorText_ = "RtApiWasapi::wasapiThread: Unable to initialize render audio client.";
- goto Exit;
- }
-
- hr = renderAudioClient->GetService( __uuidof( IAudioRenderClient ),
- ( void** ) &renderClient );
- if ( FAILED( hr ) ) {
- errorText_ = "RtApiWasapi::wasapiThread: Unable to retrieve render client handle.";
- goto Exit;
- }
-
- // configure renderEvent to trigger on every available render buffer
- renderEvent = CreateEvent( NULL, FALSE, FALSE, NULL );
- if ( !renderEvent ) {
- errorType = RtAudioError::SYSTEM_ERROR;
- errorText_ = "RtApiWasapi::wasapiThread: Unable to create render event.";
- goto Exit;
- }
-
- hr = renderAudioClient->SetEventHandle( renderEvent );
- if ( FAILED( hr ) ) {
- errorText_ = "RtApiWasapi::wasapiThread: Unable to set render event handle.";
- goto Exit;
- }
-
- ( ( WasapiHandle* ) stream_.apiHandle )->renderClient = renderClient;
- ( ( WasapiHandle* ) stream_.apiHandle )->renderEvent = renderEvent;
- }
-
- unsigned int outBufferSize = 0;
- hr = renderAudioClient->GetBufferSize( &outBufferSize );
- if ( FAILED( hr ) ) {
- errorText_ = "RtApiWasapi::wasapiThread: Unable to get render buffer size.";
- goto Exit;
- }
-
- // scale inBufferSize according to user->stream sample rate ratio
- unsigned int inBufferSize = ( unsigned int ) ( stream_.bufferSize * renderSrRatio ) * stream_.nDeviceChannels[OUTPUT];
- outBufferSize *= stream_.nDeviceChannels[OUTPUT];
-
- // set renderBuffer size
- renderBuffer.setBufferSize( inBufferSize + outBufferSize, formatBytes( stream_.deviceFormat[OUTPUT] ) );
-
- // reset the render stream
- hr = renderAudioClient->Reset();
- if ( FAILED( hr ) ) {
- errorText_ = "RtApiWasapi::wasapiThread: Unable to reset render stream.";
- goto Exit;
- }
-
- // start the render stream
- hr = renderAudioClient->Start();
- if ( FAILED( hr ) ) {
- errorText_ = "RtApiWasapi::wasapiThread: Unable to start render stream.";
- goto Exit;
- }
- }
-
- if ( stream_.mode == INPUT ) {
- convBuffSize = ( size_t ) ( stream_.bufferSize * captureSrRatio ) * stream_.nDeviceChannels[INPUT] * formatBytes( stream_.deviceFormat[INPUT] );
- deviceBuffSize = stream_.bufferSize * stream_.nDeviceChannels[INPUT] * formatBytes( stream_.deviceFormat[INPUT] );
- }
- else if ( stream_.mode == OUTPUT ) {
- convBuffSize = ( size_t ) ( stream_.bufferSize * renderSrRatio ) * stream_.nDeviceChannels[OUTPUT] * formatBytes( stream_.deviceFormat[OUTPUT] );
- deviceBuffSize = stream_.bufferSize * stream_.nDeviceChannels[OUTPUT] * formatBytes( stream_.deviceFormat[OUTPUT] );
- }
- else if ( stream_.mode == DUPLEX ) {
- convBuffSize = std::max( ( size_t ) ( stream_.bufferSize * captureSrRatio ) * stream_.nDeviceChannels[INPUT] * formatBytes( stream_.deviceFormat[INPUT] ),
- ( size_t ) ( stream_.bufferSize * renderSrRatio ) * stream_.nDeviceChannels[OUTPUT] * formatBytes( stream_.deviceFormat[OUTPUT] ) );
- deviceBuffSize = std::max( stream_.bufferSize * stream_.nDeviceChannels[INPUT] * formatBytes( stream_.deviceFormat[INPUT] ),
- stream_.bufferSize * stream_.nDeviceChannels[OUTPUT] * formatBytes( stream_.deviceFormat[OUTPUT] ) );
- }
-
- convBuffer = ( char* ) malloc( convBuffSize );
- stream_.deviceBuffer = ( char* ) malloc( deviceBuffSize );
- if ( !convBuffer || !stream_.deviceBuffer ) {
- errorType = RtAudioError::MEMORY_ERROR;
- errorText_ = "RtApiWasapi::wasapiThread: Error allocating device buffer memory.";
- goto Exit;
- }
-
- // stream process loop
- while ( stream_.state != STREAM_STOPPING ) {
- if ( !callbackPulled ) {
- // Callback Input
- // ==============
- // 1. Pull callback buffer from inputBuffer
- // 2. If 1. was successful: Convert callback buffer to user sample rate and channel count
- // Convert callback buffer to user format
-
- if ( captureAudioClient ) {
- // Pull callback buffer from inputBuffer
- callbackPulled = captureBuffer.pullBuffer( convBuffer,
- ( unsigned int ) ( stream_.bufferSize * captureSrRatio ) * stream_.nDeviceChannels[INPUT],
- stream_.deviceFormat[INPUT] );
-
- if ( callbackPulled ) {
- // Convert callback buffer to user sample rate
- convertBufferWasapi( stream_.deviceBuffer,
- convBuffer,
- stream_.nDeviceChannels[INPUT],
- captureFormat->nSamplesPerSec,
- stream_.sampleRate,
- ( unsigned int ) ( stream_.bufferSize * captureSrRatio ),
- convBufferSize,
- stream_.deviceFormat[INPUT] );
-
- if ( stream_.doConvertBuffer[INPUT] ) {
- // Convert callback buffer to user format
- convertBuffer( stream_.userBuffer[INPUT],
- stream_.deviceBuffer,
- stream_.convertInfo[INPUT] );
- }
- else {
- // no further conversion, simple copy deviceBuffer to userBuffer
- memcpy( stream_.userBuffer[INPUT],
- stream_.deviceBuffer,
- stream_.bufferSize * stream_.nUserChannels[INPUT] * formatBytes( stream_.userFormat ) );
- }
- }
- }
- else {
- // if there is no capture stream, set callbackPulled flag
- callbackPulled = true;
- }
-
- // Execute Callback
- // ================
- // 1. Execute user callback method
- // 2. Handle return value from callback
-
- // if callback has not requested the stream to stop
- if ( callbackPulled && !callbackStopped ) {
- // Execute user callback method
- callbackResult = callback( stream_.userBuffer[OUTPUT],
- stream_.userBuffer[INPUT],
- stream_.bufferSize,
- getStreamTime(),
- captureFlags & AUDCLNT_BUFFERFLAGS_DATA_DISCONTINUITY ? RTAUDIO_INPUT_OVERFLOW : 0,
- stream_.callbackInfo.userData );
-
- // Handle return value from callback
- if ( callbackResult == 1 ) {
- // instantiate a thread to stop this thread
- HANDLE threadHandle = CreateThread( NULL, 0, stopWasapiThread, this, 0, NULL );
- if ( !threadHandle ) {
- errorType = RtAudioError::THREAD_ERROR;
- errorText_ = "RtApiWasapi::wasapiThread: Unable to instantiate stream stop thread.";
- goto Exit;
- }
- else if ( !CloseHandle( threadHandle ) ) {
- errorType = RtAudioError::THREAD_ERROR;
- errorText_ = "RtApiWasapi::wasapiThread: Unable to close stream stop thread handle.";
- goto Exit;
- }
-
- callbackStopped = true;
- }
- else if ( callbackResult == 2 ) {
- // instantiate a thread to stop this thread
- HANDLE threadHandle = CreateThread( NULL, 0, abortWasapiThread, this, 0, NULL );
- if ( !threadHandle ) {
- errorType = RtAudioError::THREAD_ERROR;
- errorText_ = "RtApiWasapi::wasapiThread: Unable to instantiate stream abort thread.";
- goto Exit;
- }
- else if ( !CloseHandle( threadHandle ) ) {
- errorType = RtAudioError::THREAD_ERROR;
- errorText_ = "RtApiWasapi::wasapiThread: Unable to close stream abort thread handle.";
- goto Exit;
- }
-
- callbackStopped = true;
- }
- }
- }
-
- // Callback Output
- // ===============
- // 1. Convert callback buffer to stream format
- // 2. Convert callback buffer to stream sample rate and channel count
- // 3. Push callback buffer into outputBuffer
-
- if ( renderAudioClient && callbackPulled ) {
- if ( stream_.doConvertBuffer[OUTPUT] ) {
- // Convert callback buffer to stream format
- convertBuffer( stream_.deviceBuffer,
- stream_.userBuffer[OUTPUT],
- stream_.convertInfo[OUTPUT] );
-
- }
-
- // Convert callback buffer to stream sample rate
- convertBufferWasapi( convBuffer,
- stream_.deviceBuffer,
- stream_.nDeviceChannels[OUTPUT],
- stream_.sampleRate,
- renderFormat->nSamplesPerSec,
- stream_.bufferSize,
- convBufferSize,
- stream_.deviceFormat[OUTPUT] );
-
- // Push callback buffer into outputBuffer
- callbackPushed = renderBuffer.pushBuffer( convBuffer,
- convBufferSize * stream_.nDeviceChannels[OUTPUT],
- stream_.deviceFormat[OUTPUT] );
- }
- else {
- // if there is no render stream, set callbackPushed flag
- callbackPushed = true;
- }
-
- // Stream Capture
- // ==============
- // 1. Get capture buffer from stream
- // 2. Push capture buffer into inputBuffer
- // 3. If 2. was successful: Release capture buffer
-
- if ( captureAudioClient ) {
- // if the callback input buffer was not pulled from captureBuffer, wait for next capture event
- if ( !callbackPulled ) {
- WaitForSingleObject( captureEvent, INFINITE );
- }
-
- // Get capture buffer from stream
- hr = captureClient->GetBuffer( &streamBuffer,
- &bufferFrameCount,
- &captureFlags, NULL, NULL );
- if ( FAILED( hr ) ) {
- errorText_ = "RtApiWasapi::wasapiThread: Unable to retrieve capture buffer.";
- goto Exit;
- }
-
- if ( bufferFrameCount != 0 ) {
- // Push capture buffer into inputBuffer
- if ( captureBuffer.pushBuffer( ( char* ) streamBuffer,
- bufferFrameCount * stream_.nDeviceChannels[INPUT],
- stream_.deviceFormat[INPUT] ) )
- {
- // Release capture buffer
- hr = captureClient->ReleaseBuffer( bufferFrameCount );
- if ( FAILED( hr ) ) {
- errorText_ = "RtApiWasapi::wasapiThread: Unable to release capture buffer.";
- goto Exit;
- }
- }
- else
- {
- // Inform WASAPI that capture was unsuccessful
- hr = captureClient->ReleaseBuffer( 0 );
- if ( FAILED( hr ) ) {
- errorText_ = "RtApiWasapi::wasapiThread: Unable to release capture buffer.";
- goto Exit;
- }
- }
- }
- else
- {
- // Inform WASAPI that capture was unsuccessful
- hr = captureClient->ReleaseBuffer( 0 );
- if ( FAILED( hr ) ) {
- errorText_ = "RtApiWasapi::wasapiThread: Unable to release capture buffer.";
- goto Exit;
- }
- }
- }
-
- // Stream Render
- // =============
- // 1. Get render buffer from stream
- // 2. Pull next buffer from outputBuffer
- // 3. If 2. was successful: Fill render buffer with next buffer
- // Release render buffer
-
- if ( renderAudioClient ) {
- // if the callback output buffer was not pushed to renderBuffer, wait for next render event
- if ( callbackPulled && !callbackPushed ) {
- WaitForSingleObject( renderEvent, INFINITE );
- }
-
- // Get render buffer from stream
- hr = renderAudioClient->GetBufferSize( &bufferFrameCount );
- if ( FAILED( hr ) ) {
- errorText_ = "RtApiWasapi::wasapiThread: Unable to retrieve render buffer size.";
- goto Exit;
- }
-
- hr = renderAudioClient->GetCurrentPadding( &numFramesPadding );
- if ( FAILED( hr ) ) {
- errorText_ = "RtApiWasapi::wasapiThread: Unable to retrieve render buffer padding.";
- goto Exit;
- }
-
- bufferFrameCount -= numFramesPadding;
-
- if ( bufferFrameCount != 0 ) {
- hr = renderClient->GetBuffer( bufferFrameCount, &streamBuffer );
- if ( FAILED( hr ) ) {
- errorText_ = "RtApiWasapi::wasapiThread: Unable to retrieve render buffer.";
- goto Exit;
- }
-
- // Pull next buffer from outputBuffer
- // Fill render buffer with next buffer
- if ( renderBuffer.pullBuffer( ( char* ) streamBuffer,
- bufferFrameCount * stream_.nDeviceChannels[OUTPUT],
- stream_.deviceFormat[OUTPUT] ) )
- {
- // Release render buffer
- hr = renderClient->ReleaseBuffer( bufferFrameCount, 0 );
- if ( FAILED( hr ) ) {
- errorText_ = "RtApiWasapi::wasapiThread: Unable to release render buffer.";
- goto Exit;
- }
- }
- else
- {
- // Inform WASAPI that render was unsuccessful
- hr = renderClient->ReleaseBuffer( 0, 0 );
- if ( FAILED( hr ) ) {
- errorText_ = "RtApiWasapi::wasapiThread: Unable to release render buffer.";
- goto Exit;
- }
- }
- }
- else
- {
- // Inform WASAPI that render was unsuccessful
- hr = renderClient->ReleaseBuffer( 0, 0 );
- if ( FAILED( hr ) ) {
- errorText_ = "RtApiWasapi::wasapiThread: Unable to release render buffer.";
- goto Exit;
- }
- }
- }
-
- // if the callback buffer was pushed renderBuffer reset callbackPulled flag
- if ( callbackPushed ) {
- callbackPulled = false;
- }
-
- // tick stream time
- RtApi::tickStreamTime();
- }
-
-Exit:
- // clean up
- CoTaskMemFree( captureFormat );
- CoTaskMemFree( renderFormat );
-
- free ( convBuffer );
-
- CoUninitialize();
-
- // update stream state
- stream_.state = STREAM_STOPPED;
-
- if ( errorText_.empty() )
- return;
- else
- error( errorType );
-}
-
-//******************** End of __WINDOWS_WASAPI__ *********************//
-#endif
-
-
-#if defined(__WINDOWS_DS__) // Windows DirectSound API
-
-// 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
-#define WAVE_FORMAT_96M08 0x00010000 /* 96 kHz, Mono, 8-bit */
-#define WAVE_FORMAT_96S08 0x00020000 /* 96 kHz, Stereo, 8-bit */
-#define WAVE_FORMAT_96M16 0x00040000 /* 96 kHz, Mono, 16-bit */
-#define WAVE_FORMAT_96S16 0x00080000 /* 96 kHz, Stereo, 16-bit */
-#endif
-
-#define MINIMUM_DEVICE_BUFFER_SIZE 32768
-
-#ifdef _MSC_VER // if Microsoft Visual C++
-#pragma comment( lib, "winmm.lib" ) // then, auto-link winmm.lib. Otherwise, it has to be added manually.
-#endif
-
-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; }
-};
-
-// Declarations for utility functions, callbacks, and structures
-// specific to the DirectSound implementation.
-static BOOL CALLBACK deviceQueryCallback( LPGUID lpguid,
- LPCTSTR description,
- LPCTSTR module,
- LPVOID lpContext );
-
-static const char* getErrorString( int code );
-
-static unsigned __stdcall callbackHandler( void *ptr );
-
-struct DsDevice {
- LPGUID id[2];
- bool validId[2];
- bool found;
- std::string name;
-
- DsDevice()
- : found(false) { validId[0] = false; validId[1] = false; }
-};
-
-struct DsProbeData {
- bool isInput;
- std::vector<struct DsDevice>* dsDevices;
-};
-
-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;
-}
-
-RtApiDs :: ~RtApiDs()
-{
- if ( coInitialized_ ) CoUninitialize(); // balanced call.
- if ( stream_.state != STREAM_CLOSED ) closeStream();
-}
-
-// The DirectSound default output is always the first device.
-unsigned int RtApiDs :: getDefaultOutputDevice( void )
-{
- return 0;
-}
-
-// 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.
- struct DsProbeData probeInfo;
- probeInfo.isInput = false;
- probeInfo.dsDevices = &dsDevices;
- HRESULT result = DirectSoundEnumerate( (LPDSENUMCALLBACK) deviceQueryCallback, &probeInfo );
- if ( FAILED( result ) ) {
- errorStream_ << "RtApiDs::getDeviceCount: error (" << getErrorString( result ) << ") enumerating output devices!";
- errorText_ = errorStream_.str();
- error( RtAudioError::WARNING );
- }
-
- // Query DirectSoundCapture devices.
- probeInfo.isInput = true;
- result = DirectSoundCaptureEnumerate( (LPDSENUMCALLBACK) deviceQueryCallback, &probeInfo );
- if ( FAILED( result ) ) {
- errorStream_ << "RtApiDs::getDeviceCount: error (" << getErrorString( result ) << ") enumerating input devices!";
- errorText_ = errorStream_.str();
- error( RtAudioError::WARNING );
- }
-
- // Clean out any devices that may have disappeared (code update submitted by Eli Zehngut).
- for ( unsigned int i=0; i<dsDevices.size(); ) {
- if ( dsDevices[i].found == false ) dsDevices.erase( dsDevices.begin() + i );
- else i++;
- }
-
- return static_cast<unsigned int>(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( RtAudioError::INVALID_USE );
- return info;
- }
- }
-
- if ( device >= dsDevices.size() ) {
- errorText_ = "RtApiDs::getDeviceInfo: device ID is invalid!";
- error( RtAudioError::INVALID_USE );
- return info;
- }
-
- HRESULT result;
- if ( dsDevices[ device ].validId[0] == false ) goto probeInput;
-
- 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( RtAudioError::WARNING );
- goto probeInput;
- }
-
- 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( RtAudioError::WARNING );
- goto probeInput;
- }
-
- // 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] );
-
- if ( !info.preferredSampleRate || ( SAMPLE_RATES[k] <= 48000 && SAMPLE_RATES[k] > info.preferredSampleRate ) )
- info.preferredSampleRate = SAMPLE_RATES[k];
- }
- }
-
- // 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( RtAudioError::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( RtAudioError::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] );
- }
- std::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;
- }
-
- size_t 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();
- 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;
- }
- }
-
- // 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;
- result = DirectSoundCreate( dsDevices[ device ].id[0], &output, NULL );
- if ( FAILED( result ) ) {
- errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") opening output device (" << dsDevices[ device ].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 (" << dsDevices[ device ].name << ")!";
- errorText_ = errorStream_.str();
- return FAILURE;
- }
-
- // 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;
- }
-
- // 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;
-
- // 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;
-
- // 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 (" << dsDevices[ device ].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 (" << dsDevices[ device ].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 (" << dsDevices[ device ].name << ")!";
- errorText_ = errorStream_.str();
- return FAILURE;
- }
-
- // 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;
- }
- }
-
- // 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;
- }
-
- dsBufferSize = dsbcaps.dwBufferBytes;
-
- // 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;
- }
-
- // Zero the DS buffer
- ZeroMemory( audioPtr, dataLen );
-
- // 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;
- }
-
- 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;
- }
-
- DSCCAPS inCaps;
- inCaps.dwSize = sizeof( inCaps );
- result = input->GetCaps( &inCaps );
- if ( FAILED( result ) ) {
- input->Release();
- errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") getting input capabilities (" << dsDevices[ device ].name << ")!";
- 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;
- }
-
- // 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;
- 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.
- 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;
- }
-
- dsBufferSize = dscbcaps.dwBufferBytes;
-
- // 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.
-
- // 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;
- }
-
- // 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 (" << dsDevices[ device ].name << ")!";
- errorText_ = errorStream_.str();
- return FAILURE;
- }
-
- 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
- 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;
- }
-
- 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;
-
- 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 callback thread.
- if ( stream_.callbackInfo.isRunning == false ) {
- unsigned threadId;
- stream_.callbackInfo.isRunning = true;
- stream_.callbackInfo.object = (void *) this;
- 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;
-
- 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;
- }
-
- 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;
- }
-
- stream_.state = STREAM_CLOSED;
- return FAILURE;
-}
-
-void RtApiDs :: closeStream()
-{
- if ( stream_.state == STREAM_CLOSED ) {
- errorText_ = "RtApiDs::closeStream(): no open stream to close!";
- error( RtAudioError::WARNING );
- return;
- }
-
- // 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();
- }
- CloseHandle( 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;
- }
- }
-
- if ( stream_.deviceBuffer ) {
- free( stream_.deviceBuffer );
- stream_.deviceBuffer = 0;
- }
-
- stream_.mode = UNINITIALIZED;
- stream_.state = STREAM_CLOSED;
-}
-
-void RtApiDs :: startStream()
-{
- verifyStream();
- if ( stream_.state == STREAM_RUNNING ) {
- errorText_ = "RtApiDs::startStream(): the stream is already running!";
- error( RtAudioError::WARNING );
- return;
- }
-
- DsHandle *handle = (DsHandle *) stream_.apiHandle;
-
- // 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 );
-
- 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] );
- }
-
- 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;
- }
- }
-
- 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;
- }
- }
-
- handle->drainCounter = 0;
- handle->internalDrain = false;
- ResetEvent( handle->condition );
- stream_.state = STREAM_RUNNING;
-
- unlock:
- if ( FAILED( result ) ) error( RtAudioError::SYSTEM_ERROR );
-}
-
-void RtApiDs :: stopStream()
-{
- verifyStream();
- if ( stream_.state == STREAM_STOPPED ) {
- errorText_ = "RtApiDs::stopStream(): the stream is already stopped!";
- error( RtAudioError::WARNING );
- return;
- }
-
- 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 = 2;
- WaitForSingleObject( handle->condition, INFINITE ); // block until signaled
- }
-
- stream_.state = STREAM_STOPPED;
-
- MUTEX_LOCK( &stream_.mutex );
-
- // 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;
- }
-
- // 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;
- }
-
- // Zero the DS buffer
- ZeroMemory( audioPtr, dataLen );
-
- // 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;
- }
-
- // If we start playing again, we must begin at beginning of buffer.
- handle->bufferPointer[0] = 0;
- }
-
- if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {
- LPDIRECTSOUNDCAPTUREBUFFER buffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1];
- audioPtr = NULL;
- dataLen = 0;
-
- stream_.state = STREAM_STOPPED;
-
- if ( stream_.mode != DUPLEX )
- MUTEX_LOCK( &stream_.mutex );
-
- result = buffer->Stop();
- if ( FAILED( result ) ) {
- errorStream_ << "RtApiDs::stopStream: 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::stopStream: 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::stopStream: error (" << getErrorString( result ) << ") unlocking input buffer!";
- errorText_ = errorStream_.str();
- goto unlock;
- }
-
- // If we start recording again, we must begin at beginning of buffer.
- handle->bufferPointer[1] = 0;
- }
-
- unlock:
- timeEndPeriod( 1 ); // revert to normal scheduler frequency on lesser windows.
- MUTEX_UNLOCK( &stream_.mutex );
-
- if ( FAILED( result ) ) error( RtAudioError::SYSTEM_ERROR );
-}
-
-void RtApiDs :: abortStream()
-{
- verifyStream();
- if ( stream_.state == STREAM_STOPPED ) {
- errorText_ = "RtApiDs::abortStream(): the stream is already stopped!";
- error( RtAudioError::WARNING );
- return;
- }
-
- DsHandle *handle = (DsHandle *) stream_.apiHandle;
- handle->drainCounter = 2;
-
- stopStream();
-}
-
-void RtApiDs :: callbackEvent()
-{
- if ( stream_.state == STREAM_STOPPED || stream_.state == STREAM_STOPPING ) {
- Sleep( 50 ); // sleep 50 milliseconds
- return;
- }
-
- if ( stream_.state == STREAM_CLOSED ) {
- errorText_ = "RtApiDs::callbackEvent(): the stream is closed ... this shouldn't happen!";
- error( RtAudioError::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 ) {
-
- stream_.state = STREAM_STOPPING;
- if ( handle->internalDrain == false )
- SetEvent( handle->condition );
- else
- stopStream();
- 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;
- }
- int cbReturnValue = callback( stream_.userBuffer[0], stream_.userBuffer[1],
- stream_.bufferSize, streamTime, status, info->userData );
- if ( cbReturnValue == 2 ) {
- stream_.state = STREAM_STOPPING;
- handle->drainCounter = 2;
- abortStream();
- return;
- }
- else if ( cbReturnValue == 1 ) {
- 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;
-
- MUTEX_LOCK( &stream_.mutex );
- if ( stream_.state == STREAM_STOPPED ) {
- MUTEX_UNLOCK( &stream_.mutex );
- return;
- }
-
- 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
- // within a ms or so of correct.
-
- LPDIRECTSOUNDBUFFER dsWriteBuffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0];
- LPDIRECTSOUNDCAPTUREBUFFER dsCaptureBuffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1];
-
- DWORD startSafeWritePointer, startSafeReadPointer;
-
- result = dsWriteBuffer->GetCurrentPosition( NULL, &startSafeWritePointer );
- if ( FAILED( result ) ) {
- errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current write position!";
- errorText_ = errorStream_.str();
- MUTEX_UNLOCK( &stream_.mutex );
- error( RtAudioError::SYSTEM_ERROR );
- return;
- }
- result = dsCaptureBuffer->GetCurrentPosition( NULL, &startSafeReadPointer );
- if ( FAILED( result ) ) {
- errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current read position!";
- errorText_ = errorStream_.str();
- MUTEX_UNLOCK( &stream_.mutex );
- error( RtAudioError::SYSTEM_ERROR );
- return;
- }
- while ( true ) {
- result = dsWriteBuffer->GetCurrentPosition( NULL, &safeWritePointer );
- if ( FAILED( result ) ) {
- errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current write position!";
- errorText_ = errorStream_.str();
- MUTEX_UNLOCK( &stream_.mutex );
- error( RtAudioError::SYSTEM_ERROR );
- return;
- }
- result = dsCaptureBuffer->GetCurrentPosition( NULL, &safeReadPointer );
- if ( FAILED( result ) ) {
- errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current read position!";
- errorText_ = errorStream_.str();
- MUTEX_UNLOCK( &stream_.mutex );
- error( RtAudioError::SYSTEM_ERROR );
- return;
- }
- if ( safeWritePointer != startSafeWritePointer && safeReadPointer != startSafeReadPointer ) break;
- Sleep( 1 );
- }
-
- //assert( handle->dsBufferSize[0] == handle->dsBufferSize[1] );
-
- 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 ) {
-
- // 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();
- MUTEX_UNLOCK( &stream_.mutex );
- error( RtAudioError::SYSTEM_ERROR );
- return;
- }
- handle->bufferPointer[0] = safeWritePointer + handle->dsPointerLeadTime[0];
- if ( handle->bufferPointer[0] >= handle->dsBufferSize[0] ) handle->bufferPointer[0] -= handle->dsBufferSize[0];
- }
-
- buffersRolling = true;
- }
-
- 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 );
- }
-
- // No byte swapping necessary in DirectSound implementation.
-
- // 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 );
-
- DWORD dsBufferSize = handle->dsBufferSize[0];
- nextWritePointer = handle->bufferPointer[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 ) << ") getting current write position!";
- errorText_ = errorStream_.str();
- error( RtAudioError::SYSTEM_ERROR );
- return;
- }
-
- // 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;
-
- // Check whether the entire write region is behind the play pointer.
- if ( leadPointer >= endWrite ) break;
-
- // 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 ( 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;
- }
-
- // 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();
- MUTEX_UNLOCK( &stream_.mutex );
- error( RtAudioError::SYSTEM_ERROR );
- return;
- }
-
- // Copy our buffer into the DS buffer
- CopyMemory( buffer1, buffer, bufferSize1 );
- if ( buffer2 != NULL ) CopyMemory( buffer2, buffer+bufferSize1, bufferSize2 );
-
- // 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();
- MUTEX_UNLOCK( &stream_.mutex );
- error( RtAudioError::SYSTEM_ERROR );
- return;
- }
- nextWritePointer = ( nextWritePointer + bufferSize1 + bufferSize2 ) % dsBufferSize;
- handle->bufferPointer[0] = nextWritePointer;
- }
-
- // Don't bother draining input
- if ( handle->drainCounter ) {
- handle->drainCounter++;
- goto unlock;
- }
-
- 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();
- MUTEX_UNLOCK( &stream_.mutex );
- error( RtAudioError::SYSTEM_ERROR );
- return;
- }
-
- 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 ( safeReadPointer < endRead && stream_.callbackInfo.isRunning ) {
- // 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();
- MUTEX_UNLOCK( &stream_.mutex );
- error( RtAudioError::SYSTEM_ERROR );
- return;
- }
-
- if ( safeReadPointer < (DWORD)nextReadPointer ) safeReadPointer += dsBufferSize; // unwrap offset
- }
- }
-
- // 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();
- MUTEX_UNLOCK( &stream_.mutex );
- error( RtAudioError::SYSTEM_ERROR );
- return;
- }
-
- 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();
- MUTEX_UNLOCK( &stream_.mutex );
- error( RtAudioError::SYSTEM_ERROR );
- return;
- }
- handle->bufferPointer[1] = nextReadPointer;
-
- // 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] );
- }
-
- unlock:
- MUTEX_UNLOCK( &stream_.mutex );
- RtApi::tickStreamTime();
-}
-
-// Definitions for utility functions and callbacks
-// specific to the DirectSound implementation.
-
-static 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;
-}
-
-static BOOL CALLBACK deviceQueryCallback( LPGUID lpguid,
- LPCTSTR description,
- LPCTSTR /*module*/,
- LPVOID lpContext )
-{
- struct DsProbeData& probeInfo = *(struct DsProbeData*) lpContext;
- std::vector<struct DsDevice>& dsDevices = *probeInfo.dsDevices;
-
- HRESULT hr;
- bool validDevice = false;
- if ( probeInfo.isInput == true ) {
- DSCCAPS caps;
- LPDIRECTSOUNDCAPTURE object;
-
- 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 )
- validDevice = true;
- }
- 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 = convertCharPointerToStdString( description );
- //if ( name == "Primary Sound Driver" || name == "Primary Sound Capture Driver" )
- if ( lpguid == NULL )
- name = "Default Device";
- if ( validDevice ) {
- for ( unsigned int i=0; i<dsDevices.size(); i++ ) {
- if ( dsDevices[i].name == name ) {
- dsDevices[i].found = true;
- if ( probeInfo.isInput ) {
- dsDevices[i].id[1] = lpguid;
- dsDevices[i].validId[1] = true;
- }
- else {
- dsDevices[i].id[0] = lpguid;
- dsDevices[i].validId[0] = true;
- }
- return TRUE;
- }
- }
-
- DsDevice device;
- device.name = name;
- device.found = true;
- if ( probeInfo.isInput ) {
- device.id[1] = lpguid;
- device.validId[1] = true;
- }
- else {
- device.id[0] = lpguid;
- device.validId[0] = true;
- }
- dsDevices.push_back( device );
- }
-
- return TRUE;
-}
-
-static const char* getErrorString( int code )
-{
- switch ( code ) {
-
- case DSERR_ALLOCATED:
- return "Already allocated";
-
- case DSERR_CONTROLUNAVAIL:
- return "Control unavailable";
-
- case DSERR_INVALIDPARAM:
- return "Invalid parameter";
-
- case DSERR_INVALIDCALL:
- return "Invalid call";
-
- case DSERR_GENERIC:
- return "Generic error";
-
- case DSERR_PRIOLEVELNEEDED:
- return "Priority level needed";
-
- case DSERR_OUTOFMEMORY:
- return "Out of memory";
-
- case DSERR_BADFORMAT:
- return "The sample rate or the channel format is not supported";
-
- case DSERR_UNSUPPORTED:
- return "Not supported";
-
- case DSERR_NODRIVER:
- return "No driver";
-
- case DSERR_ALREADYINITIALIZED:
- return "Already initialized";
-
- case DSERR_NOAGGREGATION:
- return "No aggregation";
-
- case DSERR_BUFFERLOST:
- return "Buffer lost";
-
- case DSERR_OTHERAPPHASPRIO:
- return "Another application already has priority";
-
- case DSERR_UNINITIALIZED:
- return "Uninitialized";
-
- default:
- return "DirectSound unknown error";
- }
-}
-//******************** End of __WINDOWS_DS__ *********************//
-#endif
-
-
-#if defined(__LINUX_ALSA__)
-
-#include <alsa/asoundlib.h>
-#include <unistd.h>
-
- // 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_cv;
- bool runnable;
-
- AlsaHandle()
- :synchronized(false), runnable(false) { xrun[0] = false; xrun[1] = false; }
-};
-
-static void *alsaCallbackHandler( void * ptr );
-
-RtApiAlsa :: RtApiAlsa()
-{
- // Nothing to do here.
-}
-
-RtApiAlsa :: ~RtApiAlsa()
-{
- if ( stream_.state != STREAM_CLOSED ) closeStream();
-}
-
-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( RtAudioError::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( RtAudioError::WARNING );
- break;
- }
- if ( subdevice < 0 )
- break;
- nDevices++;
- }
- nextcard:
- snd_ctl_close( handle );
- snd_card_next( &card );
- }
-
- result = snd_ctl_open( &handle, "default", 0 );
- if (result == 0) {
- nDevices++;
- snd_ctl_close( handle );
- }
-
- return nDevices;
-}
-
-RtAudio::DeviceInfo RtApiAlsa :: getDeviceInfo( unsigned int device )
-{
- RtAudio::DeviceInfo info;
- info.probed = false;
-
- unsigned nDevices = 0;
- int result, subdevice, card;
- char name[64];
- snd_ctl_t *chandle;
-
- // Count cards and devices
- card = -1;
- subdevice = -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( RtAudioError::WARNING );
- goto nextcard;
- }
- 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( RtAudioError::WARNING );
- break;
- }
- if ( subdevice < 0 ) break;
- if ( nDevices == device ) {
- sprintf( name, "hw:%d,%d", card, subdevice );
- goto foundDevice;
- }
- nDevices++;
- }
- nextcard:
- snd_ctl_close( chandle );
- snd_card_next( &card );
- }
-
- result = snd_ctl_open( &chandle, "default", SND_CTL_NONBLOCK );
- if ( result == 0 ) {
- if ( nDevices == device ) {
- strcpy( name, "default" );
- goto foundDevice;
- }
- nDevices++;
- }
-
- if ( nDevices == 0 ) {
- errorText_ = "RtApiAlsa::getDeviceInfo: no devices found!";
- error( RtAudioError::INVALID_USE );
- return info;
- }
-
- if ( device >= nDevices ) {
- errorText_ = "RtApiAlsa::getDeviceInfo: device ID is invalid!";
- error( RtAudioError::INVALID_USE );
- return info;
- }
-
- 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 ) ) {
- snd_ctl_close( chandle );
- if ( device >= devices_.size() ) {
- errorText_ = "RtApiAlsa::getDeviceInfo: device ID was not present before stream was opened.";
- error( RtAudioError::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 );
-
- // First try for playback unless default device (which has subdev -1)
- stream = SND_PCM_STREAM_PLAYBACK;
- snd_pcm_info_set_stream( pcminfo, stream );
- if ( subdevice != -1 ) {
- snd_pcm_info_set_device( pcminfo, subdevice );
- snd_pcm_info_set_subdevice( pcminfo, 0 );
-
- 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( RtAudioError::WARNING );
- 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( RtAudioError::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( RtAudioError::WARNING );
- goto captureProbe;
- }
- info.outputChannels = value;
- snd_pcm_close( phandle );
-
- captureProbe:
- stream = SND_PCM_STREAM_CAPTURE;
- snd_pcm_info_set_stream( pcminfo, stream );
-
- // Now try for capture unless default device (with subdev = -1)
- if ( subdevice != -1 ) {
- 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;
- }
- }
- else
- snd_ctl_close( chandle );
-
- 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( RtAudioError::WARNING );
- 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( RtAudioError::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( RtAudioError::WARNING );
- if ( info.outputChannels == 0 ) return info;
- goto probeParameters;
- }
- info.inputChannels = value;
- snd_pcm_close( phandle );
-
- // 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;
-
- // 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 );
-
- 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( RtAudioError::WARNING );
- return info;
- }
-
- // 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( RtAudioError::WARNING );
- return info;
- }
-
- // 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.preferredSampleRate || ( SAMPLE_RATES[i] <= 48000 && SAMPLE_RATES[i] > info.preferredSampleRate ) )
- info.preferredSampleRate = 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( RtAudioError::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 ) {
- snd_pcm_close( phandle );
- errorStream_ << "RtApiAlsa::getDeviceInfo: pcm device (" << name << ") data format not supported by RtAudio.";
- errorText_ = errorStream_.str();
- error( RtAudioError::WARNING );
- return info;
- }
-
- // Get the device name
- char *cardname;
- result = snd_card_get_name( card, &cardname );
- if ( result >= 0 ) {
- sprintf( name, "hw:%s,%d", cardname, subdevice );
- free( cardname );
- }
- 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);
-#endif
-
- // 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;
-
- if ( options && options->flags & RTAUDIO_ALSA_USE_DEFAULT )
- snprintf(name, sizeof(name), "%s", "default");
- else {
- // 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++;
- }
- snd_ctl_close( chandle );
- snd_card_next( &card );
- }
-
- result = snd_ctl_open( &chandle, "default", SND_CTL_NONBLOCK );
- if ( result == 0 ) {
- if ( nDevices == device ) {
- strcpy( name, "default" );
- goto foundDevice;
- }
- nDevices++;
- }
-
- 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;
- }
- }
-
- 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();
-
- 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 )
- 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;
- }
-
- // 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 );
-#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;
- 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;
- }
-
- if ( result < 0 ) {
- snd_pcm_close( phandle );
- errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting pcm device (" << name << ") access, " << snd_strerror( result ) << ".";
- errorText_ = errorStream_.str();
- return FAILURE;
- }
-
- // 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;
- }
-
- // 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_FLOAT;
- if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat ) == 0 ) {
- stream_.deviceFormat[mode] = RTAUDIO_FLOAT32;
- 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;
- }
-
- 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.
- snd_pcm_close( phandle );
- 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;
- }
-
- // 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;
- }
-
- // 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;
-
- // 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 (or period) size.
- int dir = 0;
- 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 number, which in ALSA is referred to as the "period".
- unsigned int periods = 0;
- if ( options && options->flags & RTAUDIO_MINIMIZE_LATENCY ) periods = 2;
- if ( options && options->numberOfBuffers > 0 ) periods = options->numberOfBuffers;
- if ( periods < 2 ) periods = 4; // a fairly safe default value
- 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 ) {
- snd_pcm_close( phandle );
- errorStream_ << "RtApiAlsa::probeDeviceOpen: system error setting buffer size for duplex stream on device (" << name << ").";
- errorText_ = errorStream_.str();
- return FAILURE;
- }
-
- 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;
- }
-
-#if defined(__RTAUDIO_DEBUG__)
- 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;
- }
-
-#if defined(__RTAUDIO_DEBUG__)
- 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_cv, NULL ) ) {
- errorText_ = "RtApiAlsa::probeDeviceOpen: error initializing pthread condition variable.";
- goto error;
- }
-
- stream_.apiHandle = (void *) apiInfo;
- apiInfo->handles[0] = 0;
- apiInfo->handles[1] = 0;
- }
- else {
- apiInfo = (AlsaHandle *) stream_.apiHandle;
- }
- apiInfo->handles[mode] = phandle;
- phandle = 0;
-
- // 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;
- }
-
- 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_.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( RtAudioError::WARNING );
- }
- }
- 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 ) {
- // We previously attempted to increase the audio callback priority
- // to SCHED_RR here via the attributes. However, while no errors
- // were reported in doing so, it did not work. So, now this is
- // done in the alsaCallbackHandler function.
- stream_.callbackInfo.doRealtime = true;
- 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;
- stream_.callbackInfo.priority = priority;
- }
-#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!";
- goto error;
- }
- }
-
- return SUCCESS;
-
- error:
- if ( apiInfo ) {
- pthread_cond_destroy( &apiInfo->runnable_cv );
- 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 ( phandle) snd_pcm_close( phandle );
-
- 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;
- }
-
- stream_.state = STREAM_CLOSED;
- return FAILURE;
-}
-
-void RtApiAlsa :: closeStream()
-{
- if ( stream_.state == STREAM_CLOSED ) {
- errorText_ = "RtApiAlsa::closeStream(): no open stream to close!";
- error( RtAudioError::WARNING );
- return;
- }
-
- AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle;
- stream_.callbackInfo.isRunning = false;
- MUTEX_LOCK( &stream_.mutex );
- if ( stream_.state == STREAM_STOPPED ) {
- apiInfo->runnable = true;
- pthread_cond_signal( &apiInfo->runnable_cv );
- }
- MUTEX_UNLOCK( &stream_.mutex );
- pthread_join( stream_.callbackInfo.thread, NULL );
-
- 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 ( apiInfo ) {
- pthread_cond_destroy( &apiInfo->runnable_cv );
- 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;
- }
-
- stream_.mode = UNINITIALIZED;
- stream_.state = STREAM_CLOSED;
-}
-
-void RtApiAlsa :: startStream()
-{
- // This method calls snd_pcm_prepare if the device isn't already in that state.
-
- verifyStream();
- if ( stream_.state == STREAM_RUNNING ) {
- errorText_ = "RtApiAlsa::startStream(): the stream is already running!";
- error( RtAudioError::WARNING );
- return;
- }
-
- MUTEX_LOCK( &stream_.mutex );
-
- 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_.mode == INPUT || stream_.mode == DUPLEX ) && !apiInfo->synchronized ) {
- result = snd_pcm_drop(handle[1]); // fix to remove stale data received since device has been open
- 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;
- }
- }
- }
-
- stream_.state = STREAM_RUNNING;
-
- unlock:
- apiInfo->runnable = true;
- pthread_cond_signal( &apiInfo->runnable_cv );
- MUTEX_UNLOCK( &stream_.mutex );
-
- if ( result >= 0 ) return;
- error( RtAudioError::SYSTEM_ERROR );
-}
-
-void RtApiAlsa :: stopStream()
-{
- verifyStream();
- if ( stream_.state == STREAM_STOPPED ) {
- errorText_ = "RtApiAlsa::stopStream(): the stream is already stopped!";
- error( RtAudioError::WARNING );
- return;
- }
-
- stream_.state = STREAM_STOPPED;
- 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] );
- 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::stopStream: error stopping input pcm device, " << snd_strerror( result ) << ".";
- errorText_ = errorStream_.str();
- goto unlock;
- }
- }
-
- unlock:
- apiInfo->runnable = false; // fixes high CPU usage when stopped
- MUTEX_UNLOCK( &stream_.mutex );
-
- if ( result >= 0 ) return;
- error( RtAudioError::SYSTEM_ERROR );
-}
-
-void RtApiAlsa :: abortStream()
-{
- verifyStream();
- if ( stream_.state == STREAM_STOPPED ) {
- errorText_ = "RtApiAlsa::abortStream(): the stream is already stopped!";
- error( RtAudioError::WARNING );
- return;
- }
-
- stream_.state = STREAM_STOPPED;
- 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 ) {
- 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 == 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;
- }
- }
-
- unlock:
- apiInfo->runnable = false; // fixes high CPU usage when stopped
- MUTEX_UNLOCK( &stream_.mutex );
-
- if ( result >= 0 ) return;
- error( RtAudioError::SYSTEM_ERROR );
-}
-
-void RtApiAlsa :: callbackEvent()
-{
- AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle;
- if ( stream_.state == STREAM_STOPPED ) {
- MUTEX_LOCK( &stream_.mutex );
- while ( !apiInfo->runnable )
- pthread_cond_wait( &apiInfo->runnable_cv, &stream_.mutex );
-
- if ( stream_.state != STREAM_RUNNING ) {
- MUTEX_UNLOCK( &stream_.mutex );
- return;
- }
- MUTEX_UNLOCK( &stream_.mutex );
- }
-
- if ( stream_.state == STREAM_CLOSED ) {
- errorText_ = "RtApiAlsa::callbackEvent(): the stream is closed ... this shouldn't happen!";
- error( RtAudioError::WARNING );
- 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;
- }
- 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 );
-
- if ( doStopStream == 2 ) {
- abortStream();
- return;
- }
-
- MUTEX_LOCK( &stream_.mutex );
-
- // The state might change while waiting on a mutex.
- if ( stream_.state == STREAM_STOPPED ) goto unlock;
-
- int result;
- char *buffer;
- int channels;
- snd_pcm_t **handle;
- snd_pcm_sframes_t frames;
- RtAudioFormat format;
- handle = (snd_pcm_t **) apiInfo->handles;
-
- 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: error, current state is " << snd_pcm_state_name( state ) << ", " << snd_strerror( result ) << ".";
- errorText_ = errorStream_.str();
- }
- }
- else {
- errorStream_ << "RtApiAlsa::callbackEvent: audio read error, " << snd_strerror( result ) << ".";
- errorText_ = errorStream_.str();
- }
- error( RtAudioError::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] );
-
- // Check stream latency
- result = snd_pcm_delay( handle[1], &frames );
- if ( result == 0 && frames > 0 ) stream_.latency[1] = frames;
- }
-
- tryOutput:
-
- if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
-
- // 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;
- }
-
- // Do byte swapping if necessary.
- if ( stream_.doByteSwap[0] )
- byteSwapBuffer(buffer, stream_.bufferSize * channels, format);
-
- // 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
- errorText_ = "RtApiAlsa::callbackEvent: audio write error, underrun.";
- }
- else {
- errorStream_ << "RtApiAlsa::callbackEvent: error, current state is " << snd_pcm_state_name( state ) << ", " << snd_strerror( result ) << ".";
- errorText_ = errorStream_.str();
- }
- }
- else {
- errorStream_ << "RtApiAlsa::callbackEvent: audio write error, " << snd_strerror( result ) << ".";
- errorText_ = errorStream_.str();
- }
- error( RtAudioError::WARNING );
- goto unlock;
- }
-
- // Check stream latency
- result = snd_pcm_delay( handle[0], &frames );
- if ( result == 0 && frames > 0 ) stream_.latency[0] = frames;
- }
-
- unlock:
- MUTEX_UNLOCK( &stream_.mutex );
-
- RtApi::tickStreamTime();
- if ( doStopStream == 1 ) this->stopStream();
-}
-
-static void *alsaCallbackHandler( void *ptr )
-{
- CallbackInfo *info = (CallbackInfo *) ptr;
- RtApiAlsa *object = (RtApiAlsa *) info->object;
- bool *isRunning = &info->isRunning;
-
-#ifdef SCHED_RR // Undefined with some OSes (eg: NetBSD 1.6.x with GNU Pthread)
- if ( &info->doRealtime ) {
- pthread_t tID = pthread_self(); // ID of this thread
- sched_param prio = { info->priority }; // scheduling priority of thread
- pthread_setschedparam( tID, SCHED_RR, &prio );
- }
-#endif
-
- while ( *isRunning == true ) {
- pthread_testcancel();
- object->callbackEvent();
- }
-
- pthread_exit( NULL );
-}
-
-//******************** End of __LINUX_ALSA__ *********************//
-#endif
-
-#if defined(__LINUX_PULSE__)
-
-// Code written by Peter Meerwald, pmeerw@pmeerw.net
-// and Tristan Matthews.
-
-#include <pulse/error.h>
-#include <pulse/simple.h>
-#include <cstdio>
-
-static const unsigned int SUPPORTED_SAMPLERATES[] = { 8000, 16000, 22050, 32000,
- 44100, 48000, 96000, 0};
-
-struct rtaudio_pa_format_mapping_t {
- RtAudioFormat rtaudio_format;
- pa_sample_format_t pa_format;
-};
-
-static const rtaudio_pa_format_mapping_t supported_sampleformats[] = {
- {RTAUDIO_SINT16, PA_SAMPLE_S16LE},
- {RTAUDIO_SINT32, PA_SAMPLE_S32LE},
- {RTAUDIO_FLOAT32, PA_SAMPLE_FLOAT32LE},
- {0, PA_SAMPLE_INVALID}};
-
-struct PulseAudioHandle {
- pa_simple *s_play;
- pa_simple *s_rec;
- pthread_t thread;
- pthread_cond_t runnable_cv;
- bool runnable;
- PulseAudioHandle() : s_play(0), s_rec(0), runnable(false) { }
-};
-
-RtApiPulse::~RtApiPulse()
-{
- if ( stream_.state != STREAM_CLOSED )
- closeStream();
-}
-
-unsigned int RtApiPulse::getDeviceCount( void )
-{
- return 1;
-}
-
-RtAudio::DeviceInfo RtApiPulse::getDeviceInfo( unsigned int /*device*/ )
-{
- RtAudio::DeviceInfo info;
- info.probed = true;
- info.name = "PulseAudio";
- info.outputChannels = 2;
- info.inputChannels = 2;
- info.duplexChannels = 2;
- info.isDefaultOutput = true;
- info.isDefaultInput = true;
-
- for ( const unsigned int *sr = SUPPORTED_SAMPLERATES; *sr; ++sr )
- info.sampleRates.push_back( *sr );
-
- info.preferredSampleRate = 48000;
- info.nativeFormats = RTAUDIO_SINT16 | RTAUDIO_SINT32 | RTAUDIO_FLOAT32;
-
- return info;
-}
-
-static void *pulseaudio_callback( void * user )
-{
- CallbackInfo *cbi = static_cast<CallbackInfo *>( user );
- RtApiPulse *context = static_cast<RtApiPulse *>( cbi->object );
- volatile bool *isRunning = &cbi->isRunning;
-
- while ( *isRunning ) {
- pthread_testcancel();
- context->callbackEvent();
- }
-
- pthread_exit( NULL );
-}
-
-void RtApiPulse::closeStream( void )
-{
- PulseAudioHandle *pah = static_cast<PulseAudioHandle *>( stream_.apiHandle );
-
- stream_.callbackInfo.isRunning = false;
- if ( pah ) {
- MUTEX_LOCK( &stream_.mutex );
- if ( stream_.state == STREAM_STOPPED ) {
- pah->runnable = true;
- pthread_cond_signal( &pah->runnable_cv );
- }
- MUTEX_UNLOCK( &stream_.mutex );
-
- pthread_join( pah->thread, 0 );
- if ( pah->s_play ) {
- pa_simple_flush( pah->s_play, NULL );
- pa_simple_free( pah->s_play );
- }
- if ( pah->s_rec )
- pa_simple_free( pah->s_rec );
-
- pthread_cond_destroy( &pah->runnable_cv );
- delete pah;
- stream_.apiHandle = 0;
- }
-
- if ( stream_.userBuffer[0] ) {
- free( stream_.userBuffer[0] );
- stream_.userBuffer[0] = 0;
- }
- if ( stream_.userBuffer[1] ) {
- free( stream_.userBuffer[1] );
- stream_.userBuffer[1] = 0;
- }
-
- stream_.state = STREAM_CLOSED;
- stream_.mode = UNINITIALIZED;
-}
-
-void RtApiPulse::callbackEvent( void )
-{
- PulseAudioHandle *pah = static_cast<PulseAudioHandle *>( stream_.apiHandle );
-
- if ( stream_.state == STREAM_STOPPED ) {
- MUTEX_LOCK( &stream_.mutex );
- while ( !pah->runnable )
- pthread_cond_wait( &pah->runnable_cv, &stream_.mutex );
-
- if ( stream_.state != STREAM_RUNNING ) {
- MUTEX_UNLOCK( &stream_.mutex );
- return;
- }
- MUTEX_UNLOCK( &stream_.mutex );
- }
-
- if ( stream_.state == STREAM_CLOSED ) {
- errorText_ = "RtApiPulse::callbackEvent(): the stream is closed ... "
- "this shouldn't happen!";
- error( RtAudioError::WARNING );
- return;
- }
-
- RtAudioCallback callback = (RtAudioCallback) stream_.callbackInfo.callback;
- double streamTime = getStreamTime();
- RtAudioStreamStatus status = 0;
- int doStopStream = callback( stream_.userBuffer[OUTPUT], stream_.userBuffer[INPUT],
- stream_.bufferSize, streamTime, status,
- stream_.callbackInfo.userData );
-
- if ( doStopStream == 2 ) {
- abortStream();
- return;
- }
-
- MUTEX_LOCK( &stream_.mutex );
- void *pulse_in = stream_.doConvertBuffer[INPUT] ? stream_.deviceBuffer : stream_.userBuffer[INPUT];
- void *pulse_out = stream_.doConvertBuffer[OUTPUT] ? stream_.deviceBuffer : stream_.userBuffer[OUTPUT];
-
- if ( stream_.state != STREAM_RUNNING )
- goto unlock;
-
- int pa_error;
- size_t bytes;
- if (stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
- if ( stream_.doConvertBuffer[OUTPUT] ) {
- convertBuffer( stream_.deviceBuffer,
- stream_.userBuffer[OUTPUT],
- stream_.convertInfo[OUTPUT] );
- bytes = stream_.nDeviceChannels[OUTPUT] * stream_.bufferSize *
- formatBytes( stream_.deviceFormat[OUTPUT] );
- } else
- bytes = stream_.nUserChannels[OUTPUT] * stream_.bufferSize *
- formatBytes( stream_.userFormat );
-
- if ( pa_simple_write( pah->s_play, pulse_out, bytes, &pa_error ) < 0 ) {
- errorStream_ << "RtApiPulse::callbackEvent: audio write error, " <<
- pa_strerror( pa_error ) << ".";
- errorText_ = errorStream_.str();
- error( RtAudioError::WARNING );
- }
- }
-
- if ( stream_.mode == INPUT || stream_.mode == DUPLEX) {
- if ( stream_.doConvertBuffer[INPUT] )
- bytes = stream_.nDeviceChannels[INPUT] * stream_.bufferSize *
- formatBytes( stream_.deviceFormat[INPUT] );
- else
- bytes = stream_.nUserChannels[INPUT] * stream_.bufferSize *
- formatBytes( stream_.userFormat );
-
- if ( pa_simple_read( pah->s_rec, pulse_in, bytes, &pa_error ) < 0 ) {
- errorStream_ << "RtApiPulse::callbackEvent: audio read error, " <<
- pa_strerror( pa_error ) << ".";
- errorText_ = errorStream_.str();
- error( RtAudioError::WARNING );
- }
- if ( stream_.doConvertBuffer[INPUT] ) {
- convertBuffer( stream_.userBuffer[INPUT],
- stream_.deviceBuffer,
- stream_.convertInfo[INPUT] );
- }
- }
-
- unlock:
- MUTEX_UNLOCK( &stream_.mutex );
- RtApi::tickStreamTime();
-
- if ( doStopStream == 1 )
- stopStream();
-}
-
-void RtApiPulse::startStream( void )
-{
- PulseAudioHandle *pah = static_cast<PulseAudioHandle *>( stream_.apiHandle );
-
- if ( stream_.state == STREAM_CLOSED ) {
- errorText_ = "RtApiPulse::startStream(): the stream is not open!";
- error( RtAudioError::INVALID_USE );
- return;
- }
- if ( stream_.state == STREAM_RUNNING ) {
- errorText_ = "RtApiPulse::startStream(): the stream is already running!";
- error( RtAudioError::WARNING );
- return;
- }
-
- MUTEX_LOCK( &stream_.mutex );
-
- stream_.state = STREAM_RUNNING;
-
- pah->runnable = true;
- pthread_cond_signal( &pah->runnable_cv );
- MUTEX_UNLOCK( &stream_.mutex );
-}
-
-void RtApiPulse::stopStream( void )
-{
- PulseAudioHandle *pah = static_cast<PulseAudioHandle *>( stream_.apiHandle );
-
- if ( stream_.state == STREAM_CLOSED ) {
- errorText_ = "RtApiPulse::stopStream(): the stream is not open!";
- error( RtAudioError::INVALID_USE );
- return;
- }
- if ( stream_.state == STREAM_STOPPED ) {
- errorText_ = "RtApiPulse::stopStream(): the stream is already stopped!";
- error( RtAudioError::WARNING );
- return;
- }
-
- stream_.state = STREAM_STOPPED;
- MUTEX_LOCK( &stream_.mutex );
-
- if ( pah && pah->s_play ) {
- int pa_error;
- if ( pa_simple_drain( pah->s_play, &pa_error ) < 0 ) {
- errorStream_ << "RtApiPulse::stopStream: error draining output device, " <<
- pa_strerror( pa_error ) << ".";
- errorText_ = errorStream_.str();
- MUTEX_UNLOCK( &stream_.mutex );
- error( RtAudioError::SYSTEM_ERROR );
- return;
- }
- }
-
- stream_.state = STREAM_STOPPED;
- MUTEX_UNLOCK( &stream_.mutex );
-}
-
-void RtApiPulse::abortStream( void )
-{
- PulseAudioHandle *pah = static_cast<PulseAudioHandle*>( stream_.apiHandle );
-
- if ( stream_.state == STREAM_CLOSED ) {
- errorText_ = "RtApiPulse::abortStream(): the stream is not open!";
- error( RtAudioError::INVALID_USE );
- return;
- }
- if ( stream_.state == STREAM_STOPPED ) {
- errorText_ = "RtApiPulse::abortStream(): the stream is already stopped!";
- error( RtAudioError::WARNING );
- return;
- }
-
- stream_.state = STREAM_STOPPED;
- MUTEX_LOCK( &stream_.mutex );
-
- if ( pah && pah->s_play ) {
- int pa_error;
- if ( pa_simple_flush( pah->s_play, &pa_error ) < 0 ) {
- errorStream_ << "RtApiPulse::abortStream: error flushing output device, " <<
- pa_strerror( pa_error ) << ".";
- errorText_ = errorStream_.str();
- MUTEX_UNLOCK( &stream_.mutex );
- error( RtAudioError::SYSTEM_ERROR );
- return;
- }
- }
-
- stream_.state = STREAM_STOPPED;
- MUTEX_UNLOCK( &stream_.mutex );
-}
-
-bool RtApiPulse::probeDeviceOpen( unsigned int device, StreamMode mode,
- unsigned int channels, unsigned int firstChannel,
- unsigned int sampleRate, RtAudioFormat format,
- unsigned int *bufferSize, RtAudio::StreamOptions *options )
-{
- PulseAudioHandle *pah = 0;
- unsigned long bufferBytes = 0;
- pa_sample_spec ss;
-
- if ( device != 0 ) return false;
- if ( mode != INPUT && mode != OUTPUT ) return false;
- if ( channels != 1 && channels != 2 ) {
- errorText_ = "RtApiPulse::probeDeviceOpen: unsupported number of channels.";
- return false;
- }
- ss.channels = channels;
-
- if ( firstChannel != 0 ) return false;
-
- bool sr_found = false;
- for ( const unsigned int *sr = SUPPORTED_SAMPLERATES; *sr; ++sr ) {
- if ( sampleRate == *sr ) {
- sr_found = true;
- stream_.sampleRate = sampleRate;
- ss.rate = sampleRate;
- break;
- }
- }
- if ( !sr_found ) {
- errorText_ = "RtApiPulse::probeDeviceOpen: unsupported sample rate.";
- return false;
- }
-
- bool sf_found = 0;
- for ( const rtaudio_pa_format_mapping_t *sf = supported_sampleformats;
- sf->rtaudio_format && sf->pa_format != PA_SAMPLE_INVALID; ++sf ) {
- if ( format == sf->rtaudio_format ) {
- sf_found = true;
- stream_.userFormat = sf->rtaudio_format;
- stream_.deviceFormat[mode] = stream_.userFormat;
- ss.format = sf->pa_format;
- break;
- }
- }
- if ( !sf_found ) { // Use internal data format conversion.
- stream_.userFormat = format;
- stream_.deviceFormat[mode] = RTAUDIO_FLOAT32;
- ss.format = PA_SAMPLE_FLOAT32LE;
- }
-
- // Set other stream parameters.
- if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) stream_.userInterleaved = false;
- else stream_.userInterleaved = true;
- stream_.deviceInterleaved[mode] = true;
- stream_.nBuffers = 1;
- stream_.doByteSwap[mode] = false;
- stream_.nUserChannels[mode] = channels;
- stream_.nDeviceChannels[mode] = channels + firstChannel;
- stream_.channelOffset[mode] = 0;
- std::string streamName = "RtAudio";
-
- // 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;
-
- // 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_ = "RtApiPulse::probeDeviceOpen: error allocating user buffer memory.";
- goto error;
- }
- stream_.bufferSize = *bufferSize;
-
- 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_ = "RtApiPulse::probeDeviceOpen: error allocating device buffer memory.";
- goto error;
- }
- }
- }
-
- stream_.device[mode] = device;
-
- // Setup the buffer conversion information structure.
- if ( stream_.doConvertBuffer[mode] ) setConvertInfo( mode, firstChannel );
-
- if ( !stream_.apiHandle ) {
- PulseAudioHandle *pah = new PulseAudioHandle;
- if ( !pah ) {
- errorText_ = "RtApiPulse::probeDeviceOpen: error allocating memory for handle.";
- goto error;
- }
-
- stream_.apiHandle = pah;
- if ( pthread_cond_init( &pah->runnable_cv, NULL ) != 0 ) {
- errorText_ = "RtApiPulse::probeDeviceOpen: error creating condition variable.";
- goto error;
- }
- }
- pah = static_cast<PulseAudioHandle *>( stream_.apiHandle );
-
- int error;
- if ( options && !options->streamName.empty() ) streamName = options->streamName;
- switch ( mode ) {
- case INPUT:
- pa_buffer_attr buffer_attr;
- buffer_attr.fragsize = bufferBytes;
- buffer_attr.maxlength = -1;
-
- pah->s_rec = pa_simple_new( NULL, streamName.c_str(), PA_STREAM_RECORD, NULL, "Record", &ss, NULL, &buffer_attr, &error );
- if ( !pah->s_rec ) {
- errorText_ = "RtApiPulse::probeDeviceOpen: error connecting input to PulseAudio server.";
- goto error;
- }
- break;
- case OUTPUT:
- pah->s_play = pa_simple_new( NULL, "RtAudio", PA_STREAM_PLAYBACK, NULL, "Playback", &ss, NULL, NULL, &error );
- if ( !pah->s_play ) {
- errorText_ = "RtApiPulse::probeDeviceOpen: error connecting output to PulseAudio server.";
- goto error;
- }
- break;
- default:
- goto error;
- }
-
- if ( stream_.mode == UNINITIALIZED )
- stream_.mode = mode;
- else if ( stream_.mode == mode )
- goto error;
- else
- stream_.mode = DUPLEX;
-
- if ( !stream_.callbackInfo.isRunning ) {
- stream_.callbackInfo.object = this;
- stream_.callbackInfo.isRunning = true;
- if ( pthread_create( &pah->thread, NULL, pulseaudio_callback, (void *)&stream_.callbackInfo) != 0 ) {
- errorText_ = "RtApiPulse::probeDeviceOpen: error creating thread.";
- goto error;
- }
- }
-
- stream_.state = STREAM_STOPPED;
- return true;
-
- error:
- if ( pah && stream_.callbackInfo.isRunning ) {
- pthread_cond_destroy( &pah->runnable_cv );
- delete pah;
- 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;
-}
-
-//******************** End of __LINUX_PULSE__ *********************//
-#endif
-
-#if defined(__LINUX_OSS__)
-
-#include <unistd.h>
-#include <sys/ioctl.h>
-#include <unistd.h>
-#include <fcntl.h>
-#include <sys/soundcard.h>
-#include <errno.h>
-#include <math.h>
-
-static 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; }
-};
-
-RtApiOss :: RtApiOss()
-{
- // Nothing to do here.
-}
-
-RtApiOss :: ~RtApiOss()
-{
- if ( stream_.state != STREAM_CLOSED ) closeStream();
-}
-
-unsigned int RtApiOss :: getDeviceCount( void )
-{
- int mixerfd = open( "/dev/mixer", O_RDWR, 0 );
- if ( mixerfd == -1 ) {
- errorText_ = "RtApiOss::getDeviceCount: error opening '/dev/mixer'.";
- error( RtAudioError::WARNING );
- return 0;
- }
-
- 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( RtAudioError::WARNING );
- return 0;
- }
-
- close( mixerfd );
- return sysinfo.numaudios;
-}
-
-RtAudio::DeviceInfo RtApiOss :: getDeviceInfo( unsigned int device )
-{
- RtAudio::DeviceInfo info;
- info.probed = false;
-
- int mixerfd = open( "/dev/mixer", O_RDWR, 0 );
- if ( mixerfd == -1 ) {
- errorText_ = "RtApiOss::getDeviceInfo: error opening '/dev/mixer'.";
- error( RtAudioError::WARNING );
- return info;
- }
-
- 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( RtAudioError::WARNING );
- return info;
- }
-
- unsigned nDevices = sysinfo.numaudios;
- if ( nDevices == 0 ) {
- close( mixerfd );
- errorText_ = "RtApiOss::getDeviceInfo: no devices found!";
- error( RtAudioError::INVALID_USE );
- return info;
- }
-
- if ( device >= nDevices ) {
- close( mixerfd );
- errorText_ = "RtApiOss::getDeviceInfo: device ID is invalid!";
- error( RtAudioError::INVALID_USE );
- 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( RtAudioError::WARNING );
- return info;
- }
-
- // 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( RtAudioError::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] );
-
- if ( !info.preferredSampleRate || ( SAMPLE_RATES[k] <= 48000 && SAMPLE_RATES[k] > info.preferredSampleRate ) )
- info.preferredSampleRate = SAMPLE_RATES[k];
-
- break;
- }
- }
- }
- }
- 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] );
-
- if ( !info.preferredSampleRate || ( SAMPLE_RATES[k] <= 48000 && SAMPLE_RATES[k] > info.preferredSampleRate ) )
- info.preferredSampleRate = SAMPLE_RATES[k];
- }
- }
- }
-
- if ( info.sampleRates.size() == 0 ) {
- errorStream_ << "RtApiOss::getDeviceInfo: no supported sample rates found for device (" << ainfo.name << ").";
- errorText_ = errorStream_.str();
- error( RtAudioError::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;
- }
-
- 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;
- }
-
- 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;
- }
-
- 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;
- }
-
- // 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;
- }
-
- 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;
- }
- // 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;
-
- // 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;
- }
- }
- */
-
- // 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;
-
- // 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;
- }
- }
- 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 ( 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_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;
- }
- }
-
- 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;
- }
- }
-
- 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 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;
- }
-
- // 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;
-
- // Save buffer size (in sample frames).
- *bufferSize = ossBufferBytes / ( formatBytes(stream_.deviceFormat[mode]) * deviceChannels );
- stream_.bufferSize = *bufferSize;
-
- // 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;
- }
-
- // 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;
-
- 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;
- }
-
- // 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;
-
- // 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 ( pthread_cond_init( &handle->runnable, NULL ) ) {
- errorText_ = "RtApiOss::probeDeviceOpen: error initializing pthread condition variable.";
- goto error;
- }
-
- stream_.apiHandle = (void *) handle;
- }
- else {
- handle = (OssHandle *) stream_.apiHandle;
- }
- 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.";
- 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;
- }
- }
- }
-
- 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;
- if ( stream_.device[0] == device ) handle->id[0] = fd;
- }
- else {
- stream_.mode = mode;
-
- // 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 );
-#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, 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;
- }
-
- 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 RtApiOss :: closeStream()
-{
- if ( stream_.state == STREAM_CLOSED ) {
- errorText_ = "RtApiOss::closeStream(): no open stream to close!";
- error( RtAudioError::WARNING );
- return;
- }
-
- 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 ( 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;
- }
-
- 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;
- }
-
- 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;
- }
-
- stream_.mode = UNINITIALIZED;
- stream_.state = STREAM_CLOSED;
-}
-
-void RtApiOss :: startStream()
-{
- verifyStream();
- if ( stream_.state == STREAM_RUNNING ) {
- errorText_ = "RtApiOss::startStream(): the stream is already running!";
- error( RtAudioError::WARNING );
- return;
- }
-
- MUTEX_LOCK( &stream_.mutex );
-
- stream_.state = STREAM_RUNNING;
-
- // No need to do anything else here ... OSS automatically starts
- // when fed samples.
-
- MUTEX_UNLOCK( &stream_.mutex );
-
- OssHandle *handle = (OssHandle *) stream_.apiHandle;
- pthread_cond_signal( &handle->runnable );
-}
-
-void RtApiOss :: stopStream()
-{
- verifyStream();
- if ( stream_.state == STREAM_STOPPED ) {
- errorText_ = "RtApiOss::stopStream(): the stream is already stopped!";
- error( RtAudioError::WARNING );
- return;
- }
-
- MUTEX_LOCK( &stream_.mutex );
-
- // The state might change while waiting on a mutex.
- if ( stream_.state == STREAM_STOPPED ) {
- MUTEX_UNLOCK( &stream_.mutex );
- return;
- }
-
- int result = 0;
- OssHandle *handle = (OssHandle *) stream_.apiHandle;
- if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
-
- // Flush the output with zeros a few times.
- char *buffer;
- int samples;
- RtAudioFormat format;
-
- 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;
- }
-
- 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( RtAudioError::WARNING );
- }
- }
-
- 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_.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;
- }
- }
-
- unlock:
- stream_.state = STREAM_STOPPED;
- MUTEX_UNLOCK( &stream_.mutex );
-
- if ( result != -1 ) return;
- error( RtAudioError::SYSTEM_ERROR );
-}
-
-void RtApiOss :: abortStream()
-{
- verifyStream();
- if ( stream_.state == STREAM_STOPPED ) {
- errorText_ = "RtApiOss::abortStream(): the stream is already stopped!";
- error( RtAudioError::WARNING );
- return;
- }
-
- MUTEX_LOCK( &stream_.mutex );
-
- // The state might change while waiting on a mutex.
- if ( stream_.state == STREAM_STOPPED ) {
- MUTEX_UNLOCK( &stream_.mutex );
- return;
- }
-
- 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_.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;
- }
- }
-
- unlock:
- stream_.state = STREAM_STOPPED;
- MUTEX_UNLOCK( &stream_.mutex );
-
- if ( result != -1 ) return;
- error( RtAudioError::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 );
- }
-
- if ( stream_.state == STREAM_CLOSED ) {
- errorText_ = "RtApiOss::callbackEvent(): the stream is closed ... this shouldn't happen!";
- error( RtAudioError::WARNING );
- return;
- }
-
- // 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;
- }
-
- MUTEX_LOCK( &stream_.mutex );
-
- // The state might change while waiting on a mutex.
- if ( stream_.state == STREAM_STOPPED ) goto unlock;
-
- int result;
- char *buffer;
- int samples;
- RtAudioFormat format;
-
- if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
-
- // 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;
- }
-
- // 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 ( 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( RtAudioError::WARNING );
- // Continue on to input section.
- }
- }
-
- if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {
-
- // 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;
- }
-
- // Read samples from device.
- result = read( handle->id[1], buffer, samples * formatBytes(format) );
-
- 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( RtAudioError::WARNING );
- goto unlock;
- }
-
- // Do byte swapping if necessary.
- if ( stream_.doByteSwap[1] )
- byteSwapBuffer( buffer, samples, format );
-
- // Do buffer conversion if necessary.
- if ( stream_.doConvertBuffer[1] )
- convertBuffer( stream_.userBuffer[1], stream_.deviceBuffer, stream_.convertInfo[1] );
- }
-
- unlock:
- MUTEX_UNLOCK( &stream_.mutex );
-
- RtApi::tickStreamTime();
- if ( doStopStream == 1 ) this->stopStream();
-}
-
-static void *ossCallbackHandler( void *ptr )
-{
- CallbackInfo *info = (CallbackInfo *) ptr;
- RtApiOss *object = (RtApiOss *) info->object;
- bool *isRunning = &info->isRunning;
-
- while ( *isRunning == true ) {
- pthread_testcancel();
- object->callbackEvent();
- }
-
- pthread_exit( NULL );
-}
-
-//******************** End of __LINUX_OSS__ *********************//
-#endif
-
-
-// *************************************************** //
-//
-// Protected common (OS-independent) RtAudio methods.
-//
-// *************************************************** //
-
-// This method can be modified to control the behavior of error
-// message printing.
-void RtApi :: error( RtAudioError::Type type )
-{
- errorStream_.str(""); // clear the ostringstream
-
- RtAudioErrorCallback errorCallback = (RtAudioErrorCallback) stream_.callbackInfo.errorCallback;
- if ( errorCallback ) {
- // abortStream() can generate new error messages. Ignore them. Just keep original one.
-
- if ( firstErrorOccurred_ )
- return;
-
- firstErrorOccurred_ = true;
- const std::string errorMessage = errorText_;
-
- if ( type != RtAudioError::WARNING && stream_.state != STREAM_STOPPED) {
- stream_.callbackInfo.isRunning = false; // exit from the thread
- abortStream();
- }
-
- errorCallback( type, errorMessage );
- firstErrorOccurred_ = false;
- return;
- }
-
- if ( type == RtAudioError::WARNING && showWarnings_ == true )
- std::cerr << '\n' << errorText_ << "\n\n";
- else if ( type != RtAudioError::WARNING )
- throw( RtAudioError( errorText_, type ) );
-}
-
-void RtApi :: verifyStream()
-{
- if ( stream_.state == STREAM_CLOSED ) {
- errorText_ = "RtApi:: a stream is not open!";
- error( RtAudioError::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;
- stream_.callbackInfo.errorCallback = 0;
- 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_SINT32 || format == RTAUDIO_FLOAT32 )
- return 4;
- else if ( format == RTAUDIO_FLOAT64 )
- return 8;
- else if ( format == RTAUDIO_SINT24 )
- return 3;
- else if ( format == RTAUDIO_SINT8 )
- return 1;
-
- errorText_ = "RtApi::formatBytes: undefined format.";
- error( RtAudioError::WARNING );
-
- 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];
- }
-
- 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 {
- 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 { // 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_.bufferSize );
- stream_.convertInfo[mode].outOffset.push_back( k * stream_.bufferSize );
- stream_.convertInfo[mode].inJump = 1;
- stream_.convertInfo[mode].outJump = 1;
- }
- }
- }
-
- // 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 {
- 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 lower 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;
- }
- }
- 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;
- }
- }
- else if (info.inFormat == RTAUDIO_SINT24) {
- Int24 *in = (Int24 *)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]].asInt());
- out[info.outOffset[j]] += 0.5;
- out[info.outOffset[j]] *= scale;
- }
- 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_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.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;
- }
- }
- else if (info.inFormat == RTAUDIO_SINT24) {
- Int24 *in = (Int24 *)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]].asInt());
- out[info.outOffset[j]] += 0.5;
- out[info.outOffset[j]] *= scale;
- }
- 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_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.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) {
- Int24 *in = (Int24 *)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]].asInt();
- 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_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.outFormat == RTAUDIO_SINT24) {
- Int24 *out = (Int24 *)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]] << 16);
- //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]] << 8);
- //out[info.outOffset[j]] <<= 8;
- }
- in += info.inJump;
- out += info.outJump;
- }
- }
- else if (info.inFormat == RTAUDIO_SINT24) {
- // Channel compensation and/or (de)interleaving only.
- Int24 *in = (Int24 *)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]] >> 8);
- //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_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.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) {
- Int24 *in = (Int24 *)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]].asInt() >> 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]] = (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_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;
- }
- }
- 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) {
- Int24 *in = (Int24 *)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]].asInt() >> 16);
- }
- 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_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_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_SINT24 ) {
- for ( unsigned int i=0; i<samples; i++ ) {
- // Swap 1st and 3rd bytes.
- val = *(ptr);
- *(ptr) = *(ptr+2);
- *(ptr+2) = val;
-
- // Increment 2 more bytes.
- ptr += 2;
- }
- }
- 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
- //
- // Local Variables:
- // c-basic-offset: 2
- // indent-tabs-mode: nil
- // End:
- //
- // vim: et sts=2 sw=2
-
-#endif
+#ifdef RTAUDIO_ENABLED +/************************************************************************/ +/*! \class RtAudio + \brief Realtime audio i/o C++ classes. + + RtAudio provides a common API (Application Programming Interface) + for realtime audio input/output across Linux (native ALSA, Jack, + and OSS), Macintosh OS X (CoreAudio and Jack), and Windows + (DirectSound, ASIO and WASAPI) operating systems. + + RtAudio WWW site: http://www.music.mcgill.ca/~gary/rtaudio/ + + RtAudio: realtime audio i/o C++ classes + Copyright (c) 2001-2014 Gary P. Scavone + + Permission is hereby granted, free of charge, to any person + obtaining a copy of this software and associated documentation files + (the "Software"), to deal in the Software without restriction, + including without limitation the rights to use, copy, modify, merge, + publish, distribute, sublicense, and/or sell copies of the Software, + and to permit persons to whom the Software is furnished to do so, + subject to the following conditions: + + The above copyright notice and this permission notice shall be + included in all copies or substantial portions of the Software. + + Any person wishing to distribute modifications to the Software is + asked to send the modifications to the original developer so that + they can be incorporated into the canonical version. This is, + however, not a binding provision of this license. + + THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, + EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF + MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. + IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR + ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF + CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION + WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE. +*/ +/************************************************************************/ + +// RtAudio: Version 4.1.1 + +#include "RtAudio.h" +#include <iostream> +#include <cstdlib> +#include <cstring> +#include <climits> +#include <algorithm> + +// Static variable definitions. +const unsigned int RtApi::MAX_SAMPLE_RATES = 14; +const unsigned int RtApi::SAMPLE_RATES[] = { + 4000, 5512, 8000, 9600, 11025, 16000, 22050, + 32000, 44100, 48000, 88200, 96000, 176400, 192000 +}; + +#if defined(__WINDOWS_DS__) || defined(__WINDOWS_ASIO__) || defined(__WINDOWS_WASAPI__) +#ifdef WINRT_ENABLED + #define MUTEX_INITIALIZE(A) InitializeCriticalSectionEx(A, 0, 0) +#else + #define MUTEX_INITIALIZE(A) InitializeCriticalSection(A) +#endif + #define MUTEX_DESTROY(A) DeleteCriticalSection(A) + #define MUTEX_LOCK(A) EnterCriticalSection(A) + #define MUTEX_UNLOCK(A) LeaveCriticalSection(A) + + #include "tchar.h" + + static std::string convertCharPointerToStdString(const char *text) + { + return std::string(text); + } + + static std::string convertCharPointerToStdString(const wchar_t *text) + { + int length = WideCharToMultiByte(CP_UTF8, 0, text, -1, NULL, 0, NULL, NULL); + std::string s( length-1, '\0' ); + WideCharToMultiByte(CP_UTF8, 0, text, -1, &s[0], length, NULL, NULL); + return s; + } + +#elif defined(__LINUX_ALSA__) || defined(__LINUX_PULSE__) || defined(__UNIX_JACK__) || defined(__LINUX_OSS__) || defined(__MACOSX_CORE__) + // pthread API + #define MUTEX_INITIALIZE(A) pthread_mutex_init(A, NULL) + #define MUTEX_DESTROY(A) pthread_mutex_destroy(A) + #define MUTEX_LOCK(A) pthread_mutex_lock(A) + #define MUTEX_UNLOCK(A) pthread_mutex_unlock(A) +#else + #define MUTEX_INITIALIZE(A) abs(*A) // dummy definitions + #define MUTEX_DESTROY(A) abs(*A) // dummy definitions +#endif + +// *************************************************** // +// +// RtAudio definitions. +// +// *************************************************** // + +std::string RtAudio :: getVersion( void ) throw() +{ + return RTAUDIO_VERSION; +} + +void RtAudio :: getCompiledApi( std::vector<RtAudio::Api> &apis ) throw() +{ + apis.clear(); + + // The order here will control the order of RtAudio's API search in + // the constructor. +#if defined(__UNIX_JACK__) + apis.push_back( UNIX_JACK ); +#endif +#if defined(__LINUX_ALSA__) + apis.push_back( LINUX_ALSA ); +#endif +#if defined(__LINUX_PULSE__) + apis.push_back( LINUX_PULSE ); +#endif +#if defined(__LINUX_OSS__) + apis.push_back( LINUX_OSS ); +#endif +#if defined(__WINDOWS_ASIO__) + apis.push_back( WINDOWS_ASIO ); +#endif +#if defined(__WINDOWS_WASAPI__) + apis.push_back( WINDOWS_WASAPI ); +#endif +#if defined(__WINDOWS_DS__) + apis.push_back( WINDOWS_DS ); +#endif +#if defined(__MACOSX_CORE__) + apis.push_back( MACOSX_CORE ); +#endif +#if defined(__RTAUDIO_DUMMY__) + apis.push_back( RTAUDIO_DUMMY ); +#endif +} + +void RtAudio :: openRtApi( RtAudio::Api api ) +{ + if ( rtapi_ ) + delete rtapi_; + rtapi_ = 0; + +#if defined(__UNIX_JACK__) + if ( api == UNIX_JACK ) + rtapi_ = new RtApiJack(); +#endif +#if defined(__LINUX_ALSA__) + if ( api == LINUX_ALSA ) + rtapi_ = new RtApiAlsa(); +#endif +#if defined(__LINUX_PULSE__) + if ( api == LINUX_PULSE ) + rtapi_ = new RtApiPulse(); +#endif +#if defined(__LINUX_OSS__) + if ( api == LINUX_OSS ) + rtapi_ = new RtApiOss(); +#endif +#if defined(__WINDOWS_ASIO__) + if ( api == WINDOWS_ASIO ) + rtapi_ = new RtApiAsio(); +#endif +#if defined(__WINDOWS_WASAPI__) + if ( api == WINDOWS_WASAPI ) + rtapi_ = new RtApiWasapi(); +#endif +#if defined(__WINDOWS_DS__) + if ( api == WINDOWS_DS ) + rtapi_ = new RtApiDs(); +#endif +#if defined(__MACOSX_CORE__) + if ( api == MACOSX_CORE ) + rtapi_ = new RtApiCore(); +#endif +#if defined(__RTAUDIO_DUMMY__) + if ( api == RTAUDIO_DUMMY ) + rtapi_ = new RtApiDummy(); +#endif +} + +RtAudio :: RtAudio( RtAudio::Api api ) +{ + rtapi_ = 0; + + if ( api != UNSPECIFIED ) { + // Attempt to open the specified API. + openRtApi( api ); + if ( rtapi_ ) return; + + // No compiled support for specified API value. Issue a debug + // warning and continue as if no API was specified. + std::cerr << "\nRtAudio: no compiled support for specified API argument!\n" << std::endl; + } + + // Iterate through the compiled APIs and return as soon as we find + // one with at least one device or we reach the end of the list. + std::vector< RtAudio::Api > apis; + getCompiledApi( apis ); + for ( unsigned int i=0; i<apis.size(); i++ ) { + openRtApi( apis[i] ); + if ( rtapi_ && rtapi_->getDeviceCount() ) break; + } + + if ( rtapi_ ) return; + + // It should not be possible to get here because the preprocessor + // definition __RTAUDIO_DUMMY__ is automatically defined if no + // API-specific definitions are passed to the compiler. But just in + // case something weird happens, we'll thow an error. + std::string errorText = "\nRtAudio: no compiled API support found ... critical error!!\n\n"; + throw( RtAudioError( errorText, RtAudioError::UNSPECIFIED ) ); +} + +RtAudio :: ~RtAudio() throw() +{ + if ( rtapi_ ) + delete rtapi_; +} + +void RtAudio :: openStream( RtAudio::StreamParameters *outputParameters, + RtAudio::StreamParameters *inputParameters, + RtAudioFormat format, unsigned int sampleRate, + unsigned int *bufferFrames, + RtAudioCallback callback, void *userData, + RtAudio::StreamOptions *options, + RtAudioErrorCallback errorCallback ) +{ + return rtapi_->openStream( outputParameters, inputParameters, format, + sampleRate, bufferFrames, callback, + userData, options, errorCallback ); +} + +// *************************************************** // +// +// Public RtApi definitions (see end of file for +// private or protected utility functions). +// +// *************************************************** // + +RtApi :: RtApi() +{ + stream_.state = STREAM_CLOSED; + stream_.mode = UNINITIALIZED; + stream_.apiHandle = 0; + stream_.userBuffer[0] = 0; + stream_.userBuffer[1] = 0; + MUTEX_INITIALIZE( &stream_.mutex ); + showWarnings_ = true; + firstErrorOccurred_ = false; +} + +RtApi :: ~RtApi() +{ + MUTEX_DESTROY( &stream_.mutex ); +} + +void RtApi :: openStream( RtAudio::StreamParameters *oParams, + RtAudio::StreamParameters *iParams, + RtAudioFormat format, unsigned int sampleRate, + unsigned int *bufferFrames, + RtAudioCallback callback, void *userData, + RtAudio::StreamOptions *options, + RtAudioErrorCallback errorCallback ) +{ + if ( stream_.state != STREAM_CLOSED ) { + errorText_ = "RtApi::openStream: a stream is already open!"; + error( RtAudioError::INVALID_USE ); + return; + } + + // Clear stream information potentially left from a previously open stream. + clearStreamInfo(); + + if ( oParams && oParams->nChannels < 1 ) { + errorText_ = "RtApi::openStream: a non-NULL output StreamParameters structure cannot have an nChannels value less than one."; + error( RtAudioError::INVALID_USE ); + return; + } + + if ( iParams && iParams->nChannels < 1 ) { + errorText_ = "RtApi::openStream: a non-NULL input StreamParameters structure cannot have an nChannels value less than one."; + error( RtAudioError::INVALID_USE ); + return; + } + + if ( oParams == NULL && iParams == NULL ) { + errorText_ = "RtApi::openStream: input and output StreamParameters structures are both NULL!"; + error( RtAudioError::INVALID_USE ); + return; + } + + if ( formatBytes(format) == 0 ) { + errorText_ = "RtApi::openStream: 'format' parameter value is undefined."; + error( RtAudioError::INVALID_USE ); + return; + } + + unsigned int nDevices = getDeviceCount(); + unsigned int oChannels = 0; + if ( oParams ) { + oChannels = oParams->nChannels; + if ( oParams->deviceId >= nDevices ) { + errorText_ = "RtApi::openStream: output device parameter value is invalid."; + error( RtAudioError::INVALID_USE ); + return; + } + } + + unsigned int iChannels = 0; + if ( iParams ) { + iChannels = iParams->nChannels; + if ( iParams->deviceId >= nDevices ) { + errorText_ = "RtApi::openStream: input device parameter value is invalid."; + error( RtAudioError::INVALID_USE ); + return; + } + } + + bool result; + + if ( oChannels > 0 ) { + + result = probeDeviceOpen( oParams->deviceId, OUTPUT, oChannels, oParams->firstChannel, + sampleRate, format, bufferFrames, options ); + if ( result == false ) { + error( RtAudioError::SYSTEM_ERROR ); + return; + } + } + + if ( iChannels > 0 ) { + + result = probeDeviceOpen( iParams->deviceId, INPUT, iChannels, iParams->firstChannel, + sampleRate, format, bufferFrames, options ); + if ( result == false ) { + if ( oChannels > 0 ) closeStream(); + error( RtAudioError::SYSTEM_ERROR ); + return; + } + } + + stream_.callbackInfo.callback = (void *) callback; + stream_.callbackInfo.userData = userData; + stream_.callbackInfo.errorCallback = (void *) errorCallback; + + if ( options ) options->numberOfBuffers = stream_.nBuffers; + stream_.state = STREAM_STOPPED; +} + +unsigned int RtApi :: getDefaultInputDevice( void ) +{ + // Should be implemented in subclasses if possible. + return 0; +} + +unsigned int RtApi :: getDefaultOutputDevice( void ) +{ + // Should be implemented in subclasses if possible. + return 0; +} + +void RtApi :: closeStream( void ) +{ + // MUST be implemented in subclasses! + return; +} + +bool RtApi :: probeDeviceOpen( unsigned int /*device*/, StreamMode /*mode*/, unsigned int /*channels*/, + unsigned int /*firstChannel*/, unsigned int /*sampleRate*/, + RtAudioFormat /*format*/, unsigned int * /*bufferSize*/, + RtAudio::StreamOptions * /*options*/ ) +{ + // MUST be implemented in subclasses! + return FAILURE; +} + +void RtApi :: tickStreamTime( void ) +{ + // Subclasses that do not provide their own implementation of + // getStreamTime should call this function once per buffer I/O to + // provide basic stream time support. + + stream_.streamTime += ( stream_.bufferSize * 1.0 / stream_.sampleRate ); + +#if defined( HAVE_GETTIMEOFDAY ) + gettimeofday( &stream_.lastTickTimestamp, NULL ); +#endif +} + +long RtApi :: getStreamLatency( void ) +{ + verifyStream(); + + long totalLatency = 0; + if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) + totalLatency = stream_.latency[0]; + if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) + totalLatency += stream_.latency[1]; + + return totalLatency; +} + +double RtApi :: getStreamTime( void ) +{ + verifyStream(); + +#if defined( HAVE_GETTIMEOFDAY ) + // Return a very accurate estimate of the stream time by + // adding in the elapsed time since the last tick. + struct timeval then; + struct timeval now; + + if ( stream_.state != STREAM_RUNNING || stream_.streamTime == 0.0 ) + return stream_.streamTime; + + gettimeofday( &now, NULL ); + then = stream_.lastTickTimestamp; + return stream_.streamTime + + ((now.tv_sec + 0.000001 * now.tv_usec) - + (then.tv_sec + 0.000001 * then.tv_usec)); +#else + return stream_.streamTime; +#endif +} + +void RtApi :: setStreamTime( double time ) +{ + verifyStream(); + + if ( time >= 0.0 ) + stream_.streamTime = time; +} + +unsigned int RtApi :: getStreamSampleRate( void ) +{ + verifyStream(); + + return stream_.sampleRate; +} + + +// *************************************************** // +// +// OS/API-specific methods. +// +// *************************************************** // + +#if defined(__MACOSX_CORE__) + +// The OS X CoreAudio API is designed to use a separate callback +// procedure for each of its audio devices. A single RtAudio duplex +// stream using two different devices is supported here, though it +// cannot be guaranteed to always behave correctly because we cannot +// synchronize these two callbacks. +// +// A property listener is installed for over/underrun information. +// However, no functionality is currently provided to allow property +// listeners to trigger user handlers because it is unclear what could +// be done if a critical stream parameter (buffer size, sample rate, +// device disconnect) notification arrived. The listeners entail +// quite a bit of extra code and most likely, a user program wouldn't +// be prepared for the result anyway. However, we do provide a flag +// to the client callback function to inform of an over/underrun. + +// A structure to hold various information related to the CoreAudio API +// 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]; + char *deviceBuffer; + pthread_cond_t condition; + int drainCounter; // Tracks callback counts when draining + bool internalDrain; // Indicates if stop is initiated from callback or not. + + CoreHandle() + :deviceBuffer(0), drainCounter(0), internalDrain(false) { nStreams[0] = 1; nStreams[1] = 1; id[0] = 0; id[1] = 0; xrun[0] = false; xrun[1] = false; } +}; + +RtApiCore:: RtApiCore() +{ +#if defined( AVAILABLE_MAC_OS_X_VERSION_10_6_AND_LATER ) + // This is a largely undocumented but absolutely necessary + // requirement starting with OS-X 10.6. If not called, queries and + // updates to various audio device properties are not handled + // correctly. + CFRunLoopRef theRunLoop = NULL; + AudioObjectPropertyAddress property = { kAudioHardwarePropertyRunLoop, + kAudioObjectPropertyScopeGlobal, + kAudioObjectPropertyElementMaster }; + OSStatus result = AudioObjectSetPropertyData( kAudioObjectSystemObject, &property, 0, NULL, sizeof(CFRunLoopRef), &theRunLoop); + if ( result != noErr ) { + errorText_ = "RtApiCore::RtApiCore: error setting run loop property!"; + error( RtAudioError::WARNING ); + } +#endif +} + +RtApiCore :: ~RtApiCore() +{ + // The subclass destructor gets called before the base class + // destructor, so close an existing stream before deallocating + // apiDeviceId memory. + if ( stream_.state != STREAM_CLOSED ) closeStream(); +} + +unsigned int RtApiCore :: getDeviceCount( void ) +{ + // Find out how many audio devices there are, if any. + UInt32 dataSize; + 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( RtAudioError::WARNING ); + return 0; + } + + return dataSize / sizeof( AudioDeviceID ); +} + +unsigned int RtApiCore :: getDefaultInputDevice( void ) +{ + unsigned int nDevices = getDeviceCount(); + if ( nDevices <= 1 ) return 0; + + AudioDeviceID id; + UInt32 dataSize = sizeof( AudioDeviceID ); + 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( RtAudioError::WARNING ); + return 0; + } + + dataSize *= nDevices; + AudioDeviceID deviceList[ nDevices ]; + 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( RtAudioError::WARNING ); + return 0; + } + + for ( unsigned int i=0; i<nDevices; i++ ) + if ( id == deviceList[i] ) return i; + + errorText_ = "RtApiCore::getDefaultInputDevice: No default device found!"; + error( RtAudioError::WARNING ); + return 0; +} + +unsigned int RtApiCore :: getDefaultOutputDevice( void ) +{ + unsigned int nDevices = getDeviceCount(); + if ( nDevices <= 1 ) return 0; + + AudioDeviceID id; + UInt32 dataSize = sizeof( AudioDeviceID ); + 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( RtAudioError::WARNING ); + return 0; + } + + dataSize = sizeof( AudioDeviceID ) * nDevices; + AudioDeviceID deviceList[ nDevices ]; + 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( RtAudioError::WARNING ); + return 0; + } + + for ( unsigned int i=0; i<nDevices; i++ ) + if ( id == deviceList[i] ) return i; + + errorText_ = "RtApiCore::getDefaultOutputDevice: No default device found!"; + error( RtAudioError::WARNING ); + return 0; +} + +RtAudio::DeviceInfo RtApiCore :: getDeviceInfo( unsigned int device ) +{ + RtAudio::DeviceInfo info; + info.probed = false; + + // Get device ID + unsigned int nDevices = getDeviceCount(); + if ( nDevices == 0 ) { + errorText_ = "RtApiCore::getDeviceInfo: no devices found!"; + error( RtAudioError::INVALID_USE ); + return info; + } + + if ( device >= nDevices ) { + errorText_ = "RtApiCore::getDeviceInfo: device ID is invalid!"; + error( RtAudioError::INVALID_USE ); + return info; + } + + AudioDeviceID deviceList[ nDevices ]; + UInt32 dataSize = sizeof( AudioDeviceID ) * nDevices; + 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( RtAudioError::WARNING ); + return info; + } + + AudioDeviceID id = deviceList[ device ]; + + // Get the device name. + info.name.erase(); + 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( RtAudioError::WARNING ); + return info; + } + + //const char *mname = CFStringGetCStringPtr( cfname, CFStringGetSystemEncoding() ); + int length = CFStringGetLength(cfname); + char *mname = (char *)malloc(length * 3 + 1); +#if defined( UNICODE ) || defined( _UNICODE ) + CFStringGetCString(cfname, mname, length * 3 + 1, kCFStringEncodingUTF8); +#else + CFStringGetCString(cfname, mname, length * 3 + 1, CFStringGetSystemEncoding()); +#endif + info.name.append( (const char *)mname, strlen(mname) ); + info.name.append( ": " ); + CFRelease( cfname ); + free(mname); + + 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( RtAudioError::WARNING ); + return info; + } + + //const char *name = CFStringGetCStringPtr( cfname, CFStringGetSystemEncoding() ); + length = CFStringGetLength(cfname); + char *name = (char *)malloc(length * 3 + 1); +#if defined( UNICODE ) || defined( _UNICODE ) + CFStringGetCString(cfname, name, length * 3 + 1, kCFStringEncodingUTF8); +#else + CFStringGetCString(cfname, name, length * 3 + 1, CFStringGetSystemEncoding()); +#endif + info.name.append( (const char *)name, strlen(name) ); + CFRelease( cfname ); + free(name); + + // Get the output stream "configuration". + AudioBufferList *bufferList = nil; + 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( RtAudioError::WARNING ); + return info; + } + + // Allocate the AudioBufferList. + bufferList = (AudioBufferList *) malloc( dataSize ); + if ( bufferList == NULL ) { + errorText_ = "RtApiCore::getDeviceInfo: memory error allocating output AudioBufferList."; + error( RtAudioError::WARNING ); + return info; + } + + 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(); + error( RtAudioError::WARNING ); + return info; + } + + // Get output channel information. + unsigned int i, nStreams = bufferList->mNumberBuffers; + for ( i=0; i<nStreams; i++ ) + info.outputChannels += bufferList->mBuffers[i].mNumberChannels; + free( bufferList ); + + // Get the input stream "configuration". + 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( RtAudioError::WARNING ); + return info; + } + + // Allocate the AudioBufferList. + bufferList = (AudioBufferList *) malloc( dataSize ); + if ( bufferList == NULL ) { + errorText_ = "RtApiCore::getDeviceInfo: memory error allocating input AudioBufferList."; + error( RtAudioError::WARNING ); + return info; + } + + 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(); + error( RtAudioError::WARNING ); + return info; + } + + // Get input channel information. + nStreams = bufferList->mNumberBuffers; + for ( i=0; i<nStreams; i++ ) + info.inputChannels += bufferList->mBuffers[i].mNumberChannels; + free( bufferList ); + + // 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; + + // Probe the device sample rates. + bool isInput = false; + if ( info.outputChannels == 0 ) isInput = true; + + // Determine the supported sample rates. + 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(); + error( RtAudioError::WARNING ); + return info; + } + + UInt32 nRanges = dataSize / sizeof( AudioValueRange ); + AudioValueRange rangeList[ nRanges ]; + result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, &rangeList ); + if ( result != kAudioHardwareNoError ) { + errorStream_ << "RtApiCore::getDeviceInfo: system error (" << getErrorCode( result ) << ") getting sample rates."; + errorText_ = errorStream_.str(); + error( RtAudioError::WARNING ); + return info; + } + + // The sample rate reporting mechanism is a bit of a mystery. It + // seems that it can either return individual rates or a range of + // rates. I assume that if the min / max range values are the same, + // then that represents a single supported rate and if the min / max + // range values are different, the device supports an arbitrary + // range of values (though there might be multiple ranges, so we'll + // use the most conservative range). + Float64 minimumRate = 1.0, maximumRate = 10000000000.0; + bool haveValueRange = false; + info.sampleRates.clear(); + for ( UInt32 i=0; i<nRanges; i++ ) { + if ( rangeList[i].mMinimum == rangeList[i].mMaximum ) { + unsigned int tmpSr = (unsigned int) rangeList[i].mMinimum; + info.sampleRates.push_back( tmpSr ); + + if ( !info.preferredSampleRate || ( tmpSr <= 48000 && tmpSr > info.preferredSampleRate ) ) + info.preferredSampleRate = tmpSr; + + } else { + haveValueRange = true; + if ( rangeList[i].mMinimum > minimumRate ) minimumRate = rangeList[i].mMinimum; + if ( rangeList[i].mMaximum < maximumRate ) maximumRate = rangeList[i].mMaximum; + } + } + + if ( haveValueRange ) { + for ( unsigned int k=0; k<MAX_SAMPLE_RATES; k++ ) { + if ( SAMPLE_RATES[k] >= (unsigned int) minimumRate && SAMPLE_RATES[k] <= (unsigned int) maximumRate ) { + info.sampleRates.push_back( SAMPLE_RATES[k] ); + + if ( !info.preferredSampleRate || ( SAMPLE_RATES[k] <= 48000 && SAMPLE_RATES[k] > info.preferredSampleRate ) ) + info.preferredSampleRate = SAMPLE_RATES[k]; + } + } + } + + // Sort and remove any redundant values + std::sort( info.sampleRates.begin(), info.sampleRates.end() ); + info.sampleRates.erase( unique( info.sampleRates.begin(), info.sampleRates.end() ), info.sampleRates.end() ); + + if ( info.sampleRates.size() == 0 ) { + errorStream_ << "RtApiCore::probeDeviceInfo: No supported sample rates found for device (" << device << ")."; + errorText_ = errorStream_.str(); + error( RtAudioError::WARNING ); + return info; + } + + // CoreAudio always uses 32-bit floating point data for PCM streams. + // Thus, any other "physical" formats supported by the device are of + // no interest to the client. + info.nativeFormats = RTAUDIO_FLOAT32; + + 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; +} + +static OSStatus callbackHandler( AudioDeviceID inDevice, + const AudioTimeStamp* /*inNow*/, + const AudioBufferList* inInputData, + const AudioTimeStamp* /*inInputTime*/, + AudioBufferList* outOutputData, + const AudioTimeStamp* /*inOutputTime*/, + void* infoPointer ) +{ + CallbackInfo *info = (CallbackInfo *) infoPointer; + + RtApiCore *object = (RtApiCore *) info->object; + if ( object->callbackEvent( inDevice, inInputData, outOutputData ) == false ) + return kAudioHardwareUnspecifiedError; + else + return kAudioHardwareNoError; +} + +static OSStatus xrunListener( AudioObjectID /*inDevice*/, + UInt32 nAddresses, + const AudioObjectPropertyAddress properties[], + void* handlePointer ) +{ + CoreHandle *handle = (CoreHandle *) handlePointer; + 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 OSStatus rateListener( AudioObjectID inDevice, + UInt32 /*nAddresses*/, + const AudioObjectPropertyAddress /*properties*/[], + void* ratePointer ) +{ + Float64 *rate = (Float64 *) ratePointer; + UInt32 dataSize = sizeof( Float64 ); + AudioObjectPropertyAddress property = { kAudioDevicePropertyNominalSampleRate, + kAudioObjectPropertyScopeGlobal, + kAudioObjectPropertyElementMaster }; + AudioObjectGetPropertyData( inDevice, &property, 0, NULL, &dataSize, rate ); + return kAudioHardwareNoError; +} + +bool RtApiCore :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels, + unsigned int firstChannel, unsigned int sampleRate, + RtAudioFormat format, unsigned int *bufferSize, + RtAudio::StreamOptions *options ) +{ + // Get device ID + unsigned int nDevices = getDeviceCount(); + if ( nDevices == 0 ) { + // This should not happen because a check is made before this function is called. + errorText_ = "RtApiCore::probeDeviceOpen: no devices found!"; + return FAILURE; + } + + if ( device >= nDevices ) { + // This should not happen because a check is made before this function is called. + errorText_ = "RtApiCore::probeDeviceOpen: device ID is invalid!"; + return FAILURE; + } + + AudioDeviceID deviceList[ nDevices ]; + UInt32 dataSize = sizeof( AudioDeviceID ) * nDevices; + 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; + } + + AudioDeviceID id = deviceList[ device ]; + + // Setup for stream mode. + bool isInput = false; + if ( mode == INPUT ) { + isInput = true; + property.mScope = kAudioDevicePropertyScopeInput; + } + else + property.mScope = kAudioDevicePropertyScopeOutput; + + // Get the stream "configuration". + 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; + } + + // Allocate the AudioBufferList. + bufferList = (AudioBufferList *) malloc( dataSize ); + if ( bufferList == NULL ) { + errorText_ = "RtApiCore::probeDeviceOpen: memory error allocating AudioBufferList."; + return FAILURE; + } + + result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, bufferList ); + if (result != noErr || dataSize == 0) { + free( bufferList ); + errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting stream configuration for device (" << device << ")."; + errorText_ = errorStream_.str(); + return FAILURE; + } + + // Search for one or more streams that contain the desired number of + // channels. CoreAudio devices can have an arbitrary number of + // streams and each stream can have an arbitrary number of channels. + // For each stream, a single buffer of interleaved samples is + // provided. RtAudio prefers the use of one stream of interleaved + // data or multiple consecutive single-channel streams. However, we + // now support multiple consecutive multi-channel streams of + // interleaved data as well. + UInt32 iStream, offsetCounter = firstChannel; + UInt32 nStreams = bufferList->mNumberBuffers; + bool monoMode = false; + bool foundStream = false; + + // First check that the device supports the requested number of + // channels. + UInt32 deviceChannels = 0; + for ( iStream=0; iStream<nStreams; iStream++ ) + deviceChannels += bufferList->mBuffers[iStream].mNumberChannels; + + if ( deviceChannels < ( channels + firstChannel ) ) { + free( bufferList ); + errorStream_ << "RtApiCore::probeDeviceOpen: the device (" << device << ") does not support the requested channel count."; + errorText_ = errorStream_.str(); + return FAILURE; + } + + // Look for a single stream meeting our needs. + UInt32 firstStream, streamCount = 1, streamChannels = 0, channelOffset = 0; + for ( iStream=0; iStream<nStreams; iStream++ ) { + streamChannels = bufferList->mBuffers[iStream].mNumberChannels; + if ( streamChannels >= channels + offsetCounter ) { + firstStream = iStream; + channelOffset = offsetCounter; + foundStream = true; + break; + } + if ( streamChannels > offsetCounter ) break; + offsetCounter -= streamChannels; + } + + // If we didn't find a single stream above, then we should be able + // to meet the channel specification with multiple streams. + if ( foundStream == false ) { + monoMode = true; + offsetCounter = firstChannel; + for ( iStream=0; iStream<nStreams; iStream++ ) { + streamChannels = bufferList->mBuffers[iStream].mNumberChannels; + if ( streamChannels > offsetCounter ) break; + offsetCounter -= streamChannels; + } + + firstStream = iStream; + channelOffset = offsetCounter; + Int32 channelCounter = channels + offsetCounter - streamChannels; + + if ( streamChannels > 1 ) monoMode = false; + while ( channelCounter > 0 ) { + streamChannels = bufferList->mBuffers[++iStream].mNumberChannels; + if ( streamChannels > 1 ) monoMode = false; + channelCounter -= streamChannels; + streamCount++; + } + } + + free( bufferList ); + + // Determine the buffer size. + AudioValueRange bufferRange; + dataSize = sizeof( AudioValueRange ); + 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(); + return FAILURE; + } + + if ( bufferRange.mMinimum > *bufferSize ) *bufferSize = (unsigned long) bufferRange.mMinimum; + else if ( bufferRange.mMaximum < *bufferSize ) *bufferSize = (unsigned long) bufferRange.mMaximum; + if ( options && options->flags & RTAUDIO_MINIMIZE_LATENCY ) *bufferSize = (unsigned long) bufferRange.mMinimum; + + // Set the buffer size. For multiple streams, I'm assuming we only + // need to make this setting for the master channel. + UInt32 theSize = (UInt32) *bufferSize; + dataSize = sizeof( UInt32 ); + 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 << ")."; + errorText_ = errorStream_.str(); + return FAILURE; + } + + // If attempting to setup a duplex stream, the bufferSize parameter + // MUST be the same in both directions! + *bufferSize = theSize; + if ( stream_.mode == OUTPUT && mode == INPUT && *bufferSize != stream_.bufferSize ) { + errorStream_ << "RtApiCore::probeDeviceOpen: system error setting buffer size for duplex stream on device (" << device << ")."; + errorText_ = errorStream_.str(); + return FAILURE; + } + + stream_.bufferSize = *bufferSize; + stream_.nBuffers = 1; + + // 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; + } + } + } + + // 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 ) { + + // Set a property listener for the sample rate change + Float64 reportedRate = 0.0; + AudioObjectPropertyAddress tmp = { kAudioDevicePropertyNominalSampleRate, kAudioObjectPropertyScopeGlobal, kAudioObjectPropertyElementMaster }; + result = AudioObjectAddPropertyListener( id, &tmp, rateListener, (void *) &reportedRate ); + if ( result != noErr ) { + errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") setting sample rate property listener for device (" << device << ")."; + errorText_ = errorStream_.str(); + return FAILURE; + } + + nominalRate = (Float64) sampleRate; + result = AudioObjectSetPropertyData( id, &property, 0, NULL, dataSize, &nominalRate ); + if ( result != noErr ) { + AudioObjectRemovePropertyListener( id, &tmp, rateListener, (void *) &reportedRate ); + errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") setting sample rate for device (" << device << ")."; + errorText_ = errorStream_.str(); + return FAILURE; + } + + // Now wait until the reported nominal rate is what we just set. + UInt32 microCounter = 0; + while ( reportedRate != nominalRate ) { + microCounter += 5000; + if ( microCounter > 5000000 ) break; + usleep( 5000 ); + } + + // Remove the property listener. + AudioObjectRemovePropertyListener( id, &tmp, rateListener, (void *) &reportedRate ); + + if ( microCounter > 5000000 ) { + errorStream_ << "RtApiCore::probeDeviceOpen: timeout waiting for sample rate update for device (" << device << ")."; + errorText_ = errorStream_.str(); + return FAILURE; + } + } + + // Now set the stream format for all streams. Also, check the + // physical format of the device and change that if necessary. + AudioStreamBasicDescription description; + dataSize = sizeof( AudioStreamBasicDescription ); + property.mSelector = kAudioStreamPropertyVirtualFormat; + result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, &description ); + if ( result != noErr ) { + errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting stream format for device (" << device << ")."; + errorText_ = errorStream_.str(); + return FAILURE; + } + + // Set the sample rate and data format id. However, only make the + // change if the sample rate is not within 1.0 of the desired + // rate and the format is not linear pcm. + bool updateFormat = false; + if ( fabs( description.mSampleRate - (Float64)sampleRate ) > 1.0 ) { + description.mSampleRate = (Float64) sampleRate; + updateFormat = true; + } + + if ( description.mFormatID != kAudioFormatLinearPCM ) { + description.mFormatID = kAudioFormatLinearPCM; + updateFormat = true; + } + + if ( updateFormat ) { + result = AudioObjectSetPropertyData( id, &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(); + return FAILURE; + } + } + + // Now check the physical format. + property.mSelector = kAudioStreamPropertyPhysicalFormat; + result = AudioObjectGetPropertyData( id, &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(); + return FAILURE; + } + + //std::cout << "Current physical stream format:" << std::endl; + //std::cout << " mBitsPerChan = " << description.mBitsPerChannel << std::endl; + //std::cout << " aligned high = " << (description.mFormatFlags & kAudioFormatFlagIsAlignedHigh) << ", isPacked = " << (description.mFormatFlags & kAudioFormatFlagIsPacked) << std::endl; + //std::cout << " bytesPerFrame = " << description.mBytesPerFrame << std::endl; + //std::cout << " sample rate = " << description.mSampleRate << std::endl; + + if ( description.mFormatID != kAudioFormatLinearPCM || description.mBitsPerChannel < 16 ) { + description.mFormatID = kAudioFormatLinearPCM; + //description.mSampleRate = (Float64) sampleRate; + AudioStreamBasicDescription testDescription = description; + UInt32 formatFlags; + + // We'll try higher bit rates first and then work our way down. + std::vector< std::pair<UInt32, UInt32> > physicalFormats; + formatFlags = (description.mFormatFlags | kLinearPCMFormatFlagIsFloat) & ~kLinearPCMFormatFlagIsSignedInteger; + physicalFormats.push_back( std::pair<Float32, UInt32>( 32, formatFlags ) ); + formatFlags = (description.mFormatFlags | kLinearPCMFormatFlagIsSignedInteger | kAudioFormatFlagIsPacked) & ~kLinearPCMFormatFlagIsFloat; + physicalFormats.push_back( std::pair<Float32, UInt32>( 32, formatFlags ) ); + physicalFormats.push_back( std::pair<Float32, UInt32>( 24, formatFlags ) ); // 24-bit packed + formatFlags &= ~( kAudioFormatFlagIsPacked | kAudioFormatFlagIsAlignedHigh ); + physicalFormats.push_back( std::pair<Float32, UInt32>( 24.2, formatFlags ) ); // 24-bit in 4 bytes, aligned low + formatFlags |= kAudioFormatFlagIsAlignedHigh; + physicalFormats.push_back( std::pair<Float32, UInt32>( 24.4, formatFlags ) ); // 24-bit in 4 bytes, aligned high + formatFlags = (description.mFormatFlags | kLinearPCMFormatFlagIsSignedInteger | kAudioFormatFlagIsPacked) & ~kLinearPCMFormatFlagIsFloat; + physicalFormats.push_back( std::pair<Float32, UInt32>( 16, formatFlags ) ); + physicalFormats.push_back( std::pair<Float32, UInt32>( 8, formatFlags ) ); + + bool setPhysicalFormat = false; + for( unsigned int i=0; i<physicalFormats.size(); i++ ) { + testDescription = description; + testDescription.mBitsPerChannel = (UInt32) physicalFormats[i].first; + testDescription.mFormatFlags = physicalFormats[i].second; + if ( (24 == (UInt32)physicalFormats[i].first) && ~( physicalFormats[i].second & kAudioFormatFlagIsPacked ) ) + testDescription.mBytesPerFrame = 4 * testDescription.mChannelsPerFrame; + else + testDescription.mBytesPerFrame = testDescription.mBitsPerChannel/8 * testDescription.mChannelsPerFrame; + testDescription.mBytesPerPacket = testDescription.mBytesPerFrame * testDescription.mFramesPerPacket; + result = AudioObjectSetPropertyData( id, &property, 0, NULL, dataSize, &testDescription ); + if ( result == noErr ) { + setPhysicalFormat = true; + //std::cout << "Updated physical stream format:" << std::endl; + //std::cout << " mBitsPerChan = " << testDescription.mBitsPerChannel << std::endl; + //std::cout << " aligned high = " << (testDescription.mFormatFlags & kAudioFormatFlagIsAlignedHigh) << ", isPacked = " << (testDescription.mFormatFlags & kAudioFormatFlagIsPacked) << std::endl; + //std::cout << " bytesPerFrame = " << testDescription.mBytesPerFrame << std::endl; + //std::cout << " sample rate = " << testDescription.mSampleRate << std::endl; + break; + } + } + + if ( !setPhysicalFormat ) { + errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") setting physical data format for device (" << device << ")."; + errorText_ = errorStream_.str(); + return FAILURE; + } + } // done setting virtual/physical formats. + + // Get the stream / device latency. + UInt32 latency; + dataSize = sizeof( UInt32 ); + 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 << ")."; + errorText_ = errorStream_.str(); + error( RtAudioError::WARNING ); + } + } + + // Byte-swapping: According to AudioHardware.h, the stream data will + // always be presented in native-endian format, so we should never + // need to byte swap. + stream_.doByteSwap[mode] = false; + + // From the CoreAudio documentation, PCM data must be supplied as + // 32-bit floats. + stream_.userFormat = format; + stream_.deviceFormat[mode] = RTAUDIO_FLOAT32; + + if ( streamCount == 1 ) + stream_.nDeviceChannels[mode] = description.mChannelsPerFrame; + else // multiple streams + stream_.nDeviceChannels[mode] = channels; + stream_.nUserChannels[mode] = channels; + stream_.channelOffset[mode] = channelOffset; // offset within a CoreAudio stream + if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) stream_.userInterleaved = false; + else stream_.userInterleaved = true; + stream_.deviceInterleaved[mode] = true; + if ( monoMode == true ) 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_.nUserChannels[mode] < stream_.nDeviceChannels[mode] ) + stream_.doConvertBuffer[mode] = true; + if ( streamCount == 1 ) { + if ( stream_.nUserChannels[mode] > 1 && + stream_.userInterleaved != stream_.deviceInterleaved[mode] ) + stream_.doConvertBuffer[mode] = true; + } + else if ( monoMode && stream_.userInterleaved ) + stream_.doConvertBuffer[mode] = true; + + // Allocate our CoreHandle structure for the stream. + CoreHandle *handle = 0; + if ( stream_.apiHandle == 0 ) { + try { + handle = new CoreHandle; + } + catch ( std::bad_alloc& ) { + errorText_ = "RtApiCore::probeDeviceOpen: error allocating CoreHandle memory."; + goto error; + } + + if ( pthread_cond_init( &handle->condition, NULL ) ) { + errorText_ = "RtApiCore::probeDeviceOpen: error initializing pthread condition variable."; + goto error; + } + stream_.apiHandle = (void *) handle; + } + else + handle = (CoreHandle *) stream_.apiHandle; + handle->iStream[mode] = firstStream; + handle->nStreams[mode] = streamCount; + handle->id[mode] = id; + + // Allocate necessary internal buffers. + unsigned long bufferBytes; + bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat ); + // stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 ); + stream_.userBuffer[mode] = (char *) malloc( bufferBytes * sizeof(char) ); + memset( stream_.userBuffer[mode], 0, bufferBytes * sizeof(char) ); + if ( stream_.userBuffer[mode] == NULL ) { + errorText_ = "RtApiCore::probeDeviceOpen: error allocating user buffer memory."; + goto error; + } + + // If possible, we will make use of the CoreAudio stream buffers as + // "device buffers". However, we can't do this if using multiple + // streams. + if ( stream_.doConvertBuffer[mode] && handle->nStreams[mode] > 1 ) { + + 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_ = "RtApiCore::probeDeviceOpen: error allocating device buffer memory."; + goto error; + } + } + } + + stream_.sampleRate = sampleRate; + stream_.device[mode] = device; + stream_.state = STREAM_STOPPED; + stream_.callbackInfo.object = (void *) this; + + // Setup the buffer conversion information structure. + if ( stream_.doConvertBuffer[mode] ) { + if ( streamCount > 1 ) setConvertInfo( mode, 0 ); + else setConvertInfo( mode, channelOffset ); + } + + if ( mode == INPUT && stream_.mode == OUTPUT && stream_.device[0] == device ) + // Only one callback procedure per device. + stream_.mode = DUPLEX; + else { +#if defined( MAC_OS_X_VERSION_10_5 ) && ( MAC_OS_X_VERSION_MIN_REQUIRED >= MAC_OS_X_VERSION_10_5 ) + result = AudioDeviceCreateIOProcID( id, callbackHandler, (void *) &stream_.callbackInfo, &handle->procId[mode] ); +#else + // deprecated in favor of AudioDeviceCreateIOProcID() + result = AudioDeviceAddIOProc( id, callbackHandler, (void *) &stream_.callbackInfo ); +#endif + if ( result != noErr ) { + errorStream_ << "RtApiCore::probeDeviceOpen: system error setting callback for device (" << device << ")."; + errorText_ = errorStream_.str(); + goto error; + } + if ( stream_.mode == OUTPUT && mode == INPUT ) + stream_.mode = DUPLEX; + else + stream_.mode = mode; + } + + // Setup the device property listener for over/underload. + property.mSelector = kAudioDeviceProcessorOverload; + property.mScope = kAudioObjectPropertyScopeGlobal; + result = AudioObjectAddPropertyListener( id, &property, xrunListener, (void *) handle ); + + return SUCCESS; + + error: + if ( handle ) { + 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; + } + } + + if ( stream_.deviceBuffer ) { + free( stream_.deviceBuffer ); + stream_.deviceBuffer = 0; + } + + stream_.state = STREAM_CLOSED; + return FAILURE; +} + +void RtApiCore :: closeStream( void ) +{ + if ( stream_.state == STREAM_CLOSED ) { + errorText_ = "RtApiCore::closeStream(): no open stream to close!"; + error( RtAudioError::WARNING ); + return; + } + + CoreHandle *handle = (CoreHandle *) stream_.apiHandle; + if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { + if (handle) { + AudioObjectPropertyAddress property = { kAudioHardwarePropertyDevices, + kAudioObjectPropertyScopeGlobal, + kAudioObjectPropertyElementMaster }; + + property.mSelector = kAudioDeviceProcessorOverload; + property.mScope = kAudioObjectPropertyScopeGlobal; + if (AudioObjectRemovePropertyListener( handle->id[0], &property, xrunListener, (void *) handle ) != noErr) { + errorText_ = "RtApiCore::closeStream(): error removing property listener!"; + error( RtAudioError::WARNING ); + } + } + if ( stream_.state == STREAM_RUNNING ) + AudioDeviceStop( handle->id[0], callbackHandler ); +#if defined( MAC_OS_X_VERSION_10_5 ) && ( MAC_OS_X_VERSION_MIN_REQUIRED >= MAC_OS_X_VERSION_10_5 ) + AudioDeviceDestroyIOProcID( handle->id[0], handle->procId[0] ); +#else + // deprecated in favor of AudioDeviceDestroyIOProcID() + AudioDeviceRemoveIOProc( handle->id[0], callbackHandler ); +#endif + } + + if ( stream_.mode == INPUT || ( stream_.mode == DUPLEX && stream_.device[0] != stream_.device[1] ) ) { + if (handle) { + AudioObjectPropertyAddress property = { kAudioHardwarePropertyDevices, + kAudioObjectPropertyScopeGlobal, + kAudioObjectPropertyElementMaster }; + + property.mSelector = kAudioDeviceProcessorOverload; + property.mScope = kAudioObjectPropertyScopeGlobal; + if (AudioObjectRemovePropertyListener( handle->id[1], &property, xrunListener, (void *) handle ) != noErr) { + errorText_ = "RtApiCore::closeStream(): error removing property listener!"; + error( RtAudioError::WARNING ); + } + } + if ( stream_.state == STREAM_RUNNING ) + AudioDeviceStop( handle->id[1], callbackHandler ); +#if defined( MAC_OS_X_VERSION_10_5 ) && ( MAC_OS_X_VERSION_MIN_REQUIRED >= MAC_OS_X_VERSION_10_5 ) + AudioDeviceDestroyIOProcID( handle->id[1], handle->procId[1] ); +#else + // deprecated in favor of AudioDeviceDestroyIOProcID() + AudioDeviceRemoveIOProc( handle->id[1], callbackHandler ); +#endif + } + + 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; + } + + // Destroy pthread condition variable. + pthread_cond_destroy( &handle->condition ); + delete handle; + stream_.apiHandle = 0; + + stream_.mode = UNINITIALIZED; + stream_.state = STREAM_CLOSED; +} + +void RtApiCore :: startStream( void ) +{ + verifyStream(); + if ( stream_.state == STREAM_RUNNING ) { + errorText_ = "RtApiCore::startStream(): the stream is already running!"; + error( RtAudioError::WARNING ); + return; + } + + OSStatus result = noErr; + CoreHandle *handle = (CoreHandle *) stream_.apiHandle; + if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { + + result = AudioDeviceStart( handle->id[0], callbackHandler ); + if ( result != noErr ) { + errorStream_ << "RtApiCore::startStream: system error (" << getErrorCode( result ) << ") starting callback procedure on device (" << stream_.device[0] << ")."; + errorText_ = errorStream_.str(); + goto unlock; + } + } + + if ( stream_.mode == INPUT || + ( stream_.mode == DUPLEX && stream_.device[0] != stream_.device[1] ) ) { + + result = AudioDeviceStart( handle->id[1], callbackHandler ); + if ( result != noErr ) { + errorStream_ << "RtApiCore::startStream: system error starting input callback procedure on device (" << stream_.device[1] << ")."; + errorText_ = errorStream_.str(); + goto unlock; + } + } + + handle->drainCounter = 0; + handle->internalDrain = false; + stream_.state = STREAM_RUNNING; + + unlock: + if ( result == noErr ) return; + error( RtAudioError::SYSTEM_ERROR ); +} + +void RtApiCore :: stopStream( void ) +{ + verifyStream(); + if ( stream_.state == STREAM_STOPPED ) { + errorText_ = "RtApiCore::stopStream(): the stream is already stopped!"; + error( RtAudioError::WARNING ); + return; + } + + OSStatus result = noErr; + CoreHandle *handle = (CoreHandle *) stream_.apiHandle; + if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { + + if ( handle->drainCounter == 0 ) { + handle->drainCounter = 2; + pthread_cond_wait( &handle->condition, &stream_.mutex ); // block until signaled + } + + result = AudioDeviceStop( handle->id[0], callbackHandler ); + if ( result != noErr ) { + errorStream_ << "RtApiCore::stopStream: system error (" << getErrorCode( result ) << ") stopping callback procedure on device (" << stream_.device[0] << ")."; + errorText_ = errorStream_.str(); + goto unlock; + } + } + + if ( stream_.mode == INPUT || ( stream_.mode == DUPLEX && stream_.device[0] != stream_.device[1] ) ) { + + result = AudioDeviceStop( handle->id[1], callbackHandler ); + if ( result != noErr ) { + errorStream_ << "RtApiCore::stopStream: system error (" << getErrorCode( result ) << ") stopping input callback procedure on device (" << stream_.device[1] << ")."; + errorText_ = errorStream_.str(); + goto unlock; + } + } + + stream_.state = STREAM_STOPPED; + + unlock: + if ( result == noErr ) return; + error( RtAudioError::SYSTEM_ERROR ); +} + +void RtApiCore :: abortStream( void ) +{ + verifyStream(); + if ( stream_.state == STREAM_STOPPED ) { + errorText_ = "RtApiCore::abortStream(): the stream is already stopped!"; + error( RtAudioError::WARNING ); + return; + } + + CoreHandle *handle = (CoreHandle *) stream_.apiHandle; + handle->drainCounter = 2; + + stopStream(); +} + +// 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 better to handle it this way because the +// callbackEvent() function probably should return before the AudioDeviceStop() +// function is called. +static void *coreStopStream( void *ptr ) +{ + CallbackInfo *info = (CallbackInfo *) ptr; + RtApiCore *object = (RtApiCore *) info->object; + + object->stopStream(); + pthread_exit( NULL ); +} + +bool RtApiCore :: callbackEvent( AudioDeviceID deviceId, + const AudioBufferList *inBufferList, + const AudioBufferList *outBufferList ) +{ + if ( stream_.state == STREAM_STOPPED || stream_.state == STREAM_STOPPING ) return SUCCESS; + if ( stream_.state == STREAM_CLOSED ) { + errorText_ = "RtApiCore::callbackEvent(): the stream is closed ... this shouldn't happen!"; + error( RtAudioError::WARNING ); + return FAILURE; + } + + CallbackInfo *info = (CallbackInfo *) &stream_.callbackInfo; + CoreHandle *handle = (CoreHandle *) stream_.apiHandle; + + // Check if we were draining the stream and signal is finished. + if ( handle->drainCounter > 3 ) { + ThreadHandle threadId; + + stream_.state = STREAM_STOPPING; + if ( handle->internalDrain == true ) + pthread_create( &threadId, NULL, coreStopStream, info ); + else // external call to stopStream() + pthread_cond_signal( &handle->condition ); + return SUCCESS; + } + + AudioDeviceID outputDevice = handle->id[0]; + + // Invoke user callback to get fresh output data UNLESS we are + // draining stream or duplex mode AND the input/output devices are + // different AND this function is called for the input device. + if ( handle->drainCounter == 0 && ( stream_.mode != DUPLEX || deviceId == outputDevice ) ) { + 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; + } + + int cbReturnValue = callback( stream_.userBuffer[0], stream_.userBuffer[1], + stream_.bufferSize, streamTime, status, info->userData ); + if ( cbReturnValue == 2 ) { + stream_.state = STREAM_STOPPING; + handle->drainCounter = 2; + abortStream(); + return SUCCESS; + } + else if ( cbReturnValue == 1 ) { + handle->drainCounter = 1; + handle->internalDrain = true; + } + } + + if ( stream_.mode == OUTPUT || ( stream_.mode == DUPLEX && deviceId == outputDevice ) ) { + + if ( handle->drainCounter > 1 ) { // write zeros to the output stream + + if ( handle->nStreams[0] == 1 ) { + memset( outBufferList->mBuffers[handle->iStream[0]].mData, + 0, + outBufferList->mBuffers[handle->iStream[0]].mDataByteSize ); + } + else { // fill multiple streams with zeros + for ( unsigned int i=0; i<handle->nStreams[0]; i++ ) { + memset( outBufferList->mBuffers[handle->iStream[0]+i].mData, + 0, + outBufferList->mBuffers[handle->iStream[0]+i].mDataByteSize ); + } + } + } + else if ( handle->nStreams[0] == 1 ) { + if ( stream_.doConvertBuffer[0] ) { // convert directly to CoreAudio stream buffer + convertBuffer( (char *) outBufferList->mBuffers[handle->iStream[0]].mData, + stream_.userBuffer[0], stream_.convertInfo[0] ); + } + else { // copy from user buffer + memcpy( outBufferList->mBuffers[handle->iStream[0]].mData, + stream_.userBuffer[0], + outBufferList->mBuffers[handle->iStream[0]].mDataByteSize ); + } + } + else { // fill multiple streams + Float32 *inBuffer = (Float32 *) stream_.userBuffer[0]; + if ( stream_.doConvertBuffer[0] ) { + convertBuffer( stream_.deviceBuffer, stream_.userBuffer[0], stream_.convertInfo[0] ); + inBuffer = (Float32 *) stream_.deviceBuffer; + } + + if ( stream_.deviceInterleaved[0] == false ) { // mono mode + 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, + (void *)&inBuffer[i*stream_.bufferSize], bufferBytes ); + } + } + else { // fill multiple multi-channel streams with interleaved data + UInt32 streamChannels, channelsLeft, inJump, outJump, inOffset; + Float32 *out, *in; + + 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 and not mono mode + inChannels = stream_.nDeviceChannels[0]; + } + + if ( inInterleaved ) inOffset = 1; + else inOffset = stream_.bufferSize; + + channelsLeft = inChannels; + for ( unsigned int i=0; i<handle->nStreams[0]; i++ ) { + in = inBuffer; + out = (Float32 *) outBufferList->mBuffers[handle->iStream[0]+i].mData; + streamChannels = outBufferList->mBuffers[handle->iStream[0]+i].mNumberChannels; + + outJump = 0; + // Account for possible channel offset in first stream + if ( i == 0 && stream_.channelOffset[0] > 0 ) { + streamChannels -= stream_.channelOffset[0]; + outJump = stream_.channelOffset[0]; + out += outJump; + } + + // Account for possible unfilled channels at end of the last stream + if ( streamChannels > channelsLeft ) { + outJump = streamChannels - channelsLeft; + streamChannels = channelsLeft; + } + + // Determine input buffer offsets and skips + if ( inInterleaved ) { + inJump = inChannels; + in += inChannels - channelsLeft; + } + else { + inJump = 1; + in += (inChannels - channelsLeft) * inOffset; + } + + for ( unsigned int i=0; i<stream_.bufferSize; i++ ) { + for ( unsigned int j=0; j<streamChannels; j++ ) { + *out++ = in[j*inOffset]; + } + out += outJump; + in += inJump; + } + channelsLeft -= streamChannels; + } + } + } + } + + // Don't bother draining input + if ( handle->drainCounter ) { + handle->drainCounter++; + goto unlock; + } + + AudioDeviceID inputDevice; + inputDevice = handle->id[1]; + if ( stream_.mode == INPUT || ( stream_.mode == DUPLEX && deviceId == inputDevice ) ) { + + if ( handle->nStreams[1] == 1 ) { + if ( stream_.doConvertBuffer[1] ) { // convert directly from CoreAudio stream buffer + convertBuffer( stream_.userBuffer[1], + (char *) inBufferList->mBuffers[handle->iStream[1]].mData, + stream_.convertInfo[1] ); + } + else { // copy to user buffer + memcpy( stream_.userBuffer[1], + inBufferList->mBuffers[handle->iStream[1]].mData, + inBufferList->mBuffers[handle->iStream[1]].mDataByteSize ); + } + } + else { // read from multiple streams + Float32 *outBuffer = (Float32 *) stream_.userBuffer[1]; + if ( stream_.doConvertBuffer[1] ) outBuffer = (Float32 *) stream_.deviceBuffer; + + 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( (void *)&outBuffer[i*stream_.bufferSize], + inBufferList->mBuffers[handle->iStream[1]+i].mData, bufferBytes ); + } + } + else { // read from multiple multi-channel streams + UInt32 streamChannels, channelsLeft, inJump, outJump, outOffset; + Float32 *out, *in; + + 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 and not mono mode + outChannels = stream_.nDeviceChannels[1]; + } + + if ( outInterleaved ) outOffset = 1; + else outOffset = stream_.bufferSize; + + channelsLeft = outChannels; + for ( unsigned int i=0; i<handle->nStreams[1]; i++ ) { + out = outBuffer; + in = (Float32 *) inBufferList->mBuffers[handle->iStream[1]+i].mData; + streamChannels = inBufferList->mBuffers[handle->iStream[1]+i].mNumberChannels; + + inJump = 0; + // Account for possible channel offset in first stream + if ( i == 0 && stream_.channelOffset[1] > 0 ) { + streamChannels -= stream_.channelOffset[1]; + inJump = stream_.channelOffset[1]; + in += inJump; + } + + // Account for possible unread channels at end of the last stream + if ( streamChannels > channelsLeft ) { + inJump = streamChannels - channelsLeft; + streamChannels = channelsLeft; + } + + // Determine output buffer offsets and skips + if ( outInterleaved ) { + outJump = outChannels; + out += outChannels - channelsLeft; + } + else { + outJump = 1; + out += (outChannels - channelsLeft) * outOffset; + } + + for ( unsigned int i=0; i<stream_.bufferSize; i++ ) { + for ( unsigned int j=0; j<streamChannels; j++ ) { + out[j*outOffset] = *in++; + } + out += outJump; + in += inJump; + } + channelsLeft -= streamChannels; + } + } + + if ( stream_.doConvertBuffer[1] ) { // convert from our internal "device" buffer + convertBuffer( stream_.userBuffer[1], + stream_.deviceBuffer, + stream_.convertInfo[1] ); + } + } + } + + unlock: + //MUTEX_UNLOCK( &stream_.mutex ); + + RtApi::tickStreamTime(); + return SUCCESS; +} + +const char* RtApiCore :: getErrorCode( OSStatus code ) +{ + switch( code ) { + + case kAudioHardwareNotRunningError: + return "kAudioHardwareNotRunningError"; + + case kAudioHardwareUnspecifiedError: + return "kAudioHardwareUnspecifiedError"; + + case kAudioHardwareUnknownPropertyError: + return "kAudioHardwareUnknownPropertyError"; + + case kAudioHardwareBadPropertySizeError: + return "kAudioHardwareBadPropertySizeError"; + + case kAudioHardwareIllegalOperationError: + return "kAudioHardwareIllegalOperationError"; + + case kAudioHardwareBadObjectError: + return "kAudioHardwareBadObjectError"; + + case kAudioHardwareBadDeviceError: + return "kAudioHardwareBadDeviceError"; + + case kAudioHardwareBadStreamError: + return "kAudioHardwareBadStreamError"; + + case kAudioHardwareUnsupportedOperationError: + return "kAudioHardwareUnsupportedOperationError"; + + case kAudioDeviceUnsupportedFormatError: + return "kAudioDeviceUnsupportedFormatError"; + + case kAudioDevicePermissionsError: + return "kAudioDevicePermissionsError"; + + 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. + +#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; } +}; + +static void jackSilentError( const char * ) {}; + +RtApiJack :: RtApiJack() +{ + // Nothing to do here. +#if !defined(__RTAUDIO_DEBUG__) + // Turn off Jack's internal error reporting. + jack_set_error_function( &jackSilentError ); +#endif +} + +RtApiJack :: ~RtApiJack() +{ + if ( stream_.state != STREAM_CLOSED ) closeStream(); +} + +unsigned int RtApiJack :: getDeviceCount( void ) +{ + // See if we can become a jack client. + jack_options_t options = (jack_options_t) ( JackNoStartServer ); //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 ); + } + + jack_client_close( client ); + return nDevices; +} + +RtAudio::DeviceInfo RtApiJack :: getDeviceInfo( unsigned int device ) +{ + RtAudio::DeviceInfo info; + info.probed = false; + + jack_options_t options = (jack_options_t) ( JackNoStartServer ); //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( RtAudioError::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 >= nDevices ) { + jack_client_close( client ); + errorText_ = "RtApiJack::getDeviceInfo: device ID is invalid!"; + error( RtAudioError::INVALID_USE ); + return info; + } + + // Get the current jack server sample rate. + info.sampleRates.clear(); + + info.preferredSampleRate = jack_get_sample_rate( client ); + info.sampleRates.push_back( info.preferredSampleRate ); + + // 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 "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); + errorText_ = "RtApiJack::getDeviceInfo: error determining Jack input/output channels!"; + error( RtAudioError::WARNING ); + return info; + } + + // 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; + + // Jack always uses 32-bit floats. + info.nativeFormats = RTAUDIO_FLOAT32; + + // 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_client_close(client); + info.probed = true; + return info; +} + +static int jackCallbackHandler( jack_nframes_t nframes, void *infoPointer ) +{ + CallbackInfo *info = (CallbackInfo *) infoPointer; + + RtApiJack *object = (RtApiJack *) info->object; + if ( object->callbackEvent( (unsigned long) nframes ) == false ) return 1; + + return 0; +} + +// 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. +static void *jackCloseStream( void *ptr ) +{ + CallbackInfo *info = (CallbackInfo *) ptr; + RtApiJack *object = (RtApiJack *) info->object; + + object->closeStream(); + + pthread_exit( NULL ); +} +static void jackShutdown( void *infoPointer ) +{ + CallbackInfo *info = (CallbackInfo *) infoPointer; + RtApiJack *object = (RtApiJack *) info->object; + + // 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; + + ThreadHandle threadId; + pthread_create( &threadId, NULL, jackCloseStream, info ); + std::cerr << "\nRtApiJack: the Jack server is shutting down this client ... stream stopped and closed!!\n" << std::endl; +} + +static int jackXrun( void *infoPointer ) +{ + JackHandle *handle = (JackHandle *) infoPointer; + + 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 ); //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( RtAudioError::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 ); + } + + if ( device >= nDevices ) { + errorText_ = "RtApiJack::probeDeviceOpen: device ID is invalid!"; + return FAILURE; + } + + // 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 ); + } + + // 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; + } + + // 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; + + // Get the latency of the JACK port. + ports = jack_get_ports( client, deviceName.c_str(), NULL, flag ); + if ( ports[ firstChannel ] ) { + // Added by Ge Wang + jack_latency_callback_mode_t cbmode = (mode == INPUT ? JackCaptureLatency : JackPlaybackLatency); + // the range (usually the min and max are equal) + jack_latency_range_t latrange; latrange.min = latrange.max = 0; + // get the latency range + jack_port_get_latency_range( jack_port_by_name( client, ports[firstChannel] ), cbmode, &latrange ); + // be optimistic, use the min! + stream_.latency[mode] = latrange.min; + //stream_.latency[mode] = jack_port_get_latency( jack_port_by_name( client, ports[ firstChannel ] ) ); + } + free( ports ); + + // The jack server always uses 32-bit floating-point data. + stream_.deviceFormat[mode] = RTAUDIO_FLOAT32; + stream_.userFormat = format; + + if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) stream_.userInterleaved = false; + else stream_.userInterleaved = true; + + // Jack always uses non-interleaved buffers. + stream_.deviceInterleaved[mode] = false; + + // Jack always provides host byte-ordered data. + stream_.doByteSwap[mode] = false; + + // 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; + + 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; + } + + 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; + + // 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; + } + + if ( stream_.doConvertBuffer[mode] ) { + + 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 ( 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; + } + } + } + + // 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; + } + + stream_.device[mode] = device; + stream_.channelOffset[mode] = firstChannel; + stream_.state = STREAM_STOPPED; + stream_.callbackInfo.object = (void *) this; + + 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 ); + } + } + + // 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 ( 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] ); + + delete handle; + 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 RtApiJack :: closeStream( void ) +{ + if ( stream_.state == STREAM_CLOSED ) { + errorText_ = "RtApiJack::closeStream(): no open stream to close!"; + error( RtAudioError::WARNING ); + return; + } + + JackHandle *handle = (JackHandle *) stream_.apiHandle; + if ( handle ) { + + if ( stream_.state == STREAM_RUNNING ) + jack_deactivate( 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; + } + + 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; + } + + 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( RtAudioError::WARNING ); + return; + } + + JackHandle *handle = (JackHandle *) stream_.apiHandle; + int result = jack_activate( handle->client ); + if ( result ) { + errorText_ = "RtApiJack::startStream(): unable to activate JACK client!"; + goto unlock; + } + + 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; + } + + // 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; + } + + // 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: + if ( result == 0 ) return; + error( RtAudioError::SYSTEM_ERROR ); +} + +void RtApiJack :: stopStream( void ) +{ + verifyStream(); + if ( stream_.state == STREAM_STOPPED ) { + errorText_ = "RtApiJack::stopStream(): the stream is already stopped!"; + error( RtAudioError::WARNING ); + return; + } + + JackHandle *handle = (JackHandle *) stream_.apiHandle; + if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { + + if ( handle->drainCounter == 0 ) { + handle->drainCounter = 2; + pthread_cond_wait( &handle->condition, &stream_.mutex ); // block until signaled + } + } + + jack_deactivate( handle->client ); + stream_.state = STREAM_STOPPED; +} + +void RtApiJack :: abortStream( void ) +{ + verifyStream(); + if ( stream_.state == STREAM_STOPPED ) { + errorText_ = "RtApiJack::abortStream(): the stream is already stopped!"; + error( RtAudioError::WARNING ); + return; + } + + JackHandle *handle = (JackHandle *) stream_.apiHandle; + handle->drainCounter = 2; + + stopStream(); +} + +// 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. +static void *jackStopStream( void *ptr ) +{ + CallbackInfo *info = (CallbackInfo *) ptr; + RtApiJack *object = (RtApiJack *) info->object; + + object->stopStream(); + pthread_exit( NULL ); +} + +bool RtApiJack :: callbackEvent( unsigned long nframes ) +{ + if ( stream_.state == STREAM_STOPPED || stream_.state == STREAM_STOPPING ) return SUCCESS; + if ( stream_.state == STREAM_CLOSED ) { + errorText_ = "RtApiCore::callbackEvent(): the stream is closed ... this shouldn't happen!"; + error( RtAudioError::WARNING ); + return FAILURE; + } + if ( stream_.bufferSize != nframes ) { + errorText_ = "RtApiCore::callbackEvent(): the JACK buffer size has changed ... cannot process!"; + error( RtAudioError::WARNING ); + return FAILURE; + } + + CallbackInfo *info = (CallbackInfo *) &stream_.callbackInfo; + JackHandle *handle = (JackHandle *) stream_.apiHandle; + + // Check if we were draining the stream and signal is finished. + if ( handle->drainCounter > 3 ) { + ThreadHandle threadId; + + stream_.state = STREAM_STOPPING; + if ( handle->internalDrain == true ) + pthread_create( &threadId, NULL, jackStopStream, info ); + else + pthread_cond_signal( &handle->condition ); + return SUCCESS; + } + + // 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; + } + int cbReturnValue = callback( stream_.userBuffer[0], stream_.userBuffer[1], + stream_.bufferSize, streamTime, status, info->userData ); + if ( cbReturnValue == 2 ) { + stream_.state = STREAM_STOPPING; + handle->drainCounter = 2; + ThreadHandle id; + pthread_create( &id, NULL, jackStopStream, info ); + return SUCCESS; + } + else if ( cbReturnValue == 1 ) { + handle->drainCounter = 1; + handle->internalDrain = true; + } + } + + jack_default_audio_sample_t *jackbuffer; + unsigned long bufferBytes = nframes * sizeof( jack_default_audio_sample_t ); + if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { + + if ( handle->drainCounter > 1 ) { // write zeros to the output stream + + 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 ); + } + + } + else if ( stream_.doConvertBuffer[0] ) { + + convertBuffer( stream_.deviceBuffer, stream_.userBuffer[0], stream_.convertInfo[0] ); + + 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 ); + } + } + } + + // Don't bother draining input + if ( handle->drainCounter ) { + handle->drainCounter++; + goto unlock; + } + + if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) { + + 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 ); + } + } + } + + unlock: + RtApi::tickStreamTime(); + return SUCCESS; +} + //******************** End of __UNIX_JACK__ *********************// +#endif + +#if defined(__WINDOWS_ASIO__) // ASIO API on Windows + +// 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. + +#include "asiosys.h" +#include "asio.h" +#include "iasiothiscallresolver.h" +#include "asiodrivers.h" +#include <cmath> + +static AsioDrivers drivers; +static ASIOCallbacks asioCallbacks; +static ASIODriverInfo driverInfo; +static CallbackInfo *asioCallbackInfo; +static bool asioXRun; + +struct AsioHandle { + int drainCounter; // Tracks callback counts when draining + bool internalDrain; // Indicates if stop is initiated from callback or not. + ASIOBufferInfo *bufferInfos; + HANDLE condition; + + AsioHandle() + :drainCounter(0), internalDrain(false), bufferInfos(0) {} +}; + +// Function declarations (definitions at end of section) +static const char* getAsioErrorString( ASIOError result ); +static void sampleRateChanged( ASIOSampleRate sRate ); +static long asioMessages( long selector, long value, void* message, double* opt ); + +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( RtAudioError::WARNING ); + } + coInitialized_ = true; + + drivers.removeCurrentDriver(); + driverInfo.asioVersion = 2; + + // See note in DirectSound implementation about GetDesktopWindow(). + driverInfo.sysRef = GetForegroundWindow(); +} + +RtApiAsio :: ~RtApiAsio() +{ + if ( stream_.state != STREAM_CLOSED ) closeStream(); + if ( coInitialized_ ) CoUninitialize(); +} + +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( RtAudioError::INVALID_USE ); + return info; + } + + if ( device >= nDevices ) { + errorText_ = "RtApiAsio::getDeviceInfo: device ID is invalid!"; + error( RtAudioError::INVALID_USE ); + return info; + } + + // 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( RtAudioError::WARNING ); + return info; + } + return devices_[ device ]; + } + + 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( RtAudioError::WARNING ); + return info; + } + + info.name = driverName; + + if ( !drivers.loadDriver( driverName ) ) { + errorStream_ << "RtApiAsio::getDeviceInfo: unable to load driver (" << driverName << ")."; + errorText_ = errorStream_.str(); + error( RtAudioError::WARNING ); + return info; + } + + result = ASIOInit( &driverInfo ); + if ( result != ASE_OK ) { + errorStream_ << "RtApiAsio::getDeviceInfo: error (" << getAsioErrorString( result ) << ") initializing driver (" << driverName << ")."; + errorText_ = errorStream_.str(); + error( RtAudioError::WARNING ); + return info; + } + + // 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( RtAudioError::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; + + // 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] ); + + if ( !info.preferredSampleRate || ( SAMPLE_RATES[i] <= 48000 && SAMPLE_RATES[i] > info.preferredSampleRate ) ) + info.preferredSampleRate = 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( RtAudioError::WARNING ); + return info; + } + + 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; + else if ( channelInfo.type == ASIOSTInt24MSB || channelInfo.type == ASIOSTInt24LSB ) + info.nativeFormats |= RTAUDIO_SINT24; + + if ( info.outputChannels > 0 ) + if ( getDefaultOutputDevice() == device ) info.isDefaultOutput = true; + if ( info.inputChannels > 0 ) + if ( getDefaultInputDevice() == device ) info.isDefaultInput = true; + + info.probed = true; + drivers.removeCurrentDriver(); + return info; +} + +static void bufferSwitch( long index, ASIOBool /*processNow*/ ) +{ + RtApiAsio *object = (RtApiAsio *) asioCallbackInfo->object; + object->callbackEvent( index ); +} + +void RtApiAsio :: saveDeviceInfo( void ) +{ + devices_.clear(); + + unsigned int nDevices = getDeviceCount(); + devices_.resize( nDevices ); + for ( unsigned int i=0; i<nDevices; i++ ) + devices_[i] = getDeviceInfo( i ); +} + +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 ) +{//////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////// + + bool isDuplexInput = mode == INPUT && stream_.mode == OUTPUT; + + // For ASIO, a duplex stream MUST use the same driver. + if ( isDuplexInput && stream_.device[0] != device ) { + errorText_ = "RtApiAsio::probeDeviceOpen: an ASIO duplex stream must use the same device for input and output!"; + return FAILURE; + } + + 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; + } + + // Only load the driver once for duplex stream. + if ( !isDuplexInput ) { + // 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(); + + if ( !drivers.loadDriver( driverName ) ) { + errorStream_ << "RtApiAsio::probeDeviceOpen: unable to load driver (" << driverName << ")."; + errorText_ = errorStream_.str(); + return FAILURE; + } + + result = ASIOInit( &driverInfo ); + if ( result != ASE_OK ) { + errorStream_ << "RtApiAsio::probeDeviceOpen: error (" << getAsioErrorString( result ) << ") initializing driver (" << driverName << ")."; + errorText_ = errorStream_.str(); + return FAILURE; + } + } + + // keep them before any "goto error", they are used for error cleanup + goto device boundary checks + bool buffersAllocated = false; + AsioHandle *handle = (AsioHandle *) stream_.apiHandle; + unsigned int nChannels; + + + // Check the device channel count. + long inputChannels, outputChannels; + result = ASIOGetChannels( &inputChannels, &outputChannels ); + if ( result != ASE_OK ) { + errorStream_ << "RtApiAsio::probeDeviceOpen: error (" << getAsioErrorString( result ) << ") getting channel count (" << driverName << ")."; + errorText_ = errorStream_.str(); + goto error; + } + + if ( ( mode == OUTPUT && (channels+firstChannel) > (unsigned int) outputChannels) || + ( mode == INPUT && (channels+firstChannel) > (unsigned int) inputChannels) ) { + errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") does not support requested channel count (" << channels << ") + offset (" << firstChannel << ")."; + errorText_ = errorStream_.str(); + goto error; + } + stream_.nDeviceChannels[mode] = channels; + stream_.nUserChannels[mode] = channels; + stream_.channelOffset[mode] = firstChannel; + + // Verify the sample rate is supported. + result = ASIOCanSampleRate( (ASIOSampleRate) sampleRate ); + if ( result != ASE_OK ) { + errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") does not support requested sample rate (" << sampleRate << ")."; + errorText_ = errorStream_.str(); + goto error; + } + + // Get the current sample rate + ASIOSampleRate currentRate; + result = ASIOGetSampleRate( ¤tRate ); + if ( result != ASE_OK ) { + errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error getting sample rate."; + errorText_ = errorStream_.str(); + goto error; + } + + // Set the sample rate only if necessary + if ( currentRate != sampleRate ) { + result = ASIOSetSampleRate( (ASIOSampleRate) sampleRate ); + if ( result != ASE_OK ) { + errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error setting sample rate (" << sampleRate << ")."; + errorText_ = errorStream_.str(); + goto error; + } + } + + // 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 ) { + errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error (" << getAsioErrorString( result ) << ") getting data format."; + errorText_ = errorStream_.str(); + goto error; + } + + // 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; + } + else if ( channelInfo.type == ASIOSTInt24MSB || channelInfo.type == ASIOSTInt24LSB ) { + stream_.deviceFormat[mode] = RTAUDIO_SINT24; + if ( channelInfo.type == ASIOSTInt24MSB ) stream_.doByteSwap[mode] = true; + } + + if ( stream_.deviceFormat[mode] == 0 ) { + errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") data format not supported by RtAudio."; + errorText_ = errorStream_.str(); + goto error; + } + + // 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 ) { + errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error (" << getAsioErrorString( result ) << ") getting buffer size."; + errorText_ = errorStream_.str(); + goto error; + } + + if ( isDuplexInput ) { + // When this is the duplex input (output was opened before), then we have to use the same + // buffersize as the output, because it might use the preferred buffer size, which most + // likely wasn't passed as input to this. The buffer sizes have to be identically anyway, + // So instead of throwing an error, make them equal. The caller uses the reference + // to the "bufferSize" param as usual to set up processing buffers. + + *bufferSize = stream_.bufferSize; + + } else { + if ( *bufferSize == 0 ) *bufferSize = preferSize; + else 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; + + 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; + } + } + + /* + // we don't use it anymore, see above! + // Just left it here for the case... + if ( isDuplexInput && stream_.bufferSize != *bufferSize ) { + errorText_ = "RtApiAsio::probeDeviceOpen: input/output buffersize discrepancy!"; + goto error; + } + */ + + stream_.bufferSize = *bufferSize; + stream_.nBuffers = 2; + + if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) stream_.userInterleaved = false; + else stream_.userInterleaved = true; + + // ASIO always uses non-interleaved buffers. + stream_.deviceInterleaved[mode] = false; + + // Allocate, if necessary, our AsioHandle structure for the stream. + if ( handle == 0 ) { + try { + handle = new AsioHandle; + } + catch ( std::bad_alloc& ) { + errorText_ = "RtApiAsio::probeDeviceOpen: error allocating AsioHandle memory."; + goto error; + } + 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. + if ( mode == INPUT && stream_.mode == OUTPUT ) { + ASIODisposeBuffers(); + if ( handle->bufferInfos ) free( handle->bufferInfos ); + } + + // Allocate, initialize, and save the bufferInfos in our stream callbackInfo structure. + 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; + } + + 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; + } + + // prepare for callbacks + stream_.sampleRate = sampleRate; + stream_.device[mode] = device; + stream_.mode = isDuplexInput ? DUPLEX : mode; + + // store this class instance before registering callbacks, that are going to use it + asioCallbackInfo = &stream_.callbackInfo; + stream_.callbackInfo.object = (void *) this; + + // 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 ) { + // Standard method failed. This can happen with strict/misbehaving drivers that return valid buffer size ranges + // but only accept the preferred buffer size as parameter for ASIOCreateBuffers. eg. Creatives ASIO driver + // in that case, let's be naïve and try that instead + *bufferSize = preferSize; + stream_.bufferSize = *bufferSize; + 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; + stream_.state = STREAM_STOPPED; + + // 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; + } + + if ( stream_.doConvertBuffer[mode] ) { + + bool makeBuffer = true; + bufferBytes = stream_.nDeviceChannels[mode] * formatBytes( stream_.deviceFormat[mode] ); + if ( isDuplexInput && 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_ = "RtApiAsio::probeDeviceOpen: error allocating device buffer memory."; + goto error; + } + } + } + + // Determine device latencies + long inputLatency, outputLatency; + result = ASIOGetLatencies( &inputLatency, &outputLatency ); + if ( result != ASE_OK ) { + errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error (" << getAsioErrorString( result ) << ") getting latency."; + errorText_ = errorStream_.str(); + error( RtAudioError::WARNING); // warn but don't fail + } + else { + stream_.latency[0] = outputLatency; + stream_.latency[1] = inputLatency; + } + + // 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 ( !isDuplexInput ) { + // the cleanup for error in the duplex input, is done by RtApi::openStream + // So we clean up for single channel only + + if ( buffersAllocated ) + ASIODisposeBuffers(); + + drivers.removeCurrentDriver(); + + if ( handle ) { + CloseHandle( handle->condition ); + if ( handle->bufferInfos ) + free( handle->bufferInfos ); + + delete handle; + stream_.apiHandle = 0; + } + + + if ( stream_.userBuffer[mode] ) { + free( stream_.userBuffer[mode] ); + stream_.userBuffer[mode] = 0; + } + + if ( stream_.deviceBuffer ) { + free( stream_.deviceBuffer ); + stream_.deviceBuffer = 0; + } + } + + return FAILURE; +}//////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////// + +void RtApiAsio :: closeStream() +{ + if ( stream_.state == STREAM_CLOSED ) { + errorText_ = "RtApiAsio::closeStream(): no open stream to close!"; + error( RtAudioError::WARNING ); + return; + } + + if ( stream_.state == STREAM_RUNNING ) { + stream_.state = STREAM_STOPPED; + ASIOStop(); + } + ASIODisposeBuffers(); + drivers.removeCurrentDriver(); + + AsioHandle *handle = (AsioHandle *) stream_.apiHandle; + if ( handle ) { + CloseHandle( handle->condition ); + if ( handle->bufferInfos ) + free( handle->bufferInfos ); + delete handle; + 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; + } + + stream_.mode = UNINITIALIZED; + stream_.state = STREAM_CLOSED; +} + +bool stopThreadCalled = false; + +void RtApiAsio :: startStream() +{ + verifyStream(); + if ( stream_.state == STREAM_RUNNING ) { + errorText_ = "RtApiAsio::startStream(): the stream is already running!"; + error( RtAudioError::WARNING ); + return; + } + + 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; + } + + handle->drainCounter = 0; + handle->internalDrain = false; + ResetEvent( handle->condition ); + stream_.state = STREAM_RUNNING; + asioXRun = false; + + unlock: + stopThreadCalled = false; + + if ( result == ASE_OK ) return; + error( RtAudioError::SYSTEM_ERROR ); +} + +void RtApiAsio :: stopStream() +{ + verifyStream(); + if ( stream_.state == STREAM_STOPPED ) { + errorText_ = "RtApiAsio::stopStream(): the stream is already stopped!"; + error( RtAudioError::WARNING ); + return; + } + + AsioHandle *handle = (AsioHandle *) stream_.apiHandle; + if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { + if ( handle->drainCounter == 0 ) { + handle->drainCounter = 2; + WaitForSingleObject( handle->condition, INFINITE ); // block until signaled + } + } + + stream_.state = STREAM_STOPPED; + + ASIOError result = ASIOStop(); + if ( result != ASE_OK ) { + errorStream_ << "RtApiAsio::stopStream: error (" << getAsioErrorString( result ) << ") stopping device."; + errorText_ = errorStream_.str(); + } + + if ( result == ASE_OK ) return; + error( RtAudioError::SYSTEM_ERROR ); +} + +void RtApiAsio :: abortStream() +{ + verifyStream(); + if ( stream_.state == STREAM_STOPPED ) { + errorText_ = "RtApiAsio::abortStream(): the stream is already stopped!"; + error( RtAudioError::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 = 2; + stopStream(); +} + +// 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 ASIOStop() +// function will return. +static unsigned __stdcall asioStopStream( void *ptr ) +{ + CallbackInfo *info = (CallbackInfo *) ptr; + RtApiAsio *object = (RtApiAsio *) info->object; + + object->stopStream(); + _endthreadex( 0 ); + return 0; +} + +bool RtApiAsio :: callbackEvent( long bufferIndex ) +{ + if ( stream_.state == STREAM_STOPPED || stream_.state == STREAM_STOPPING ) return SUCCESS; + if ( stream_.state == STREAM_CLOSED ) { + errorText_ = "RtApiAsio::callbackEvent(): the stream is closed ... this shouldn't happen!"; + error( RtAudioError::WARNING ); + return FAILURE; + } + + CallbackInfo *info = (CallbackInfo *) &stream_.callbackInfo; + AsioHandle *handle = (AsioHandle *) stream_.apiHandle; + + // Check if we were draining the stream and signal if finished. + if ( handle->drainCounter > 3 ) { + + stream_.state = STREAM_STOPPING; + if ( handle->internalDrain == false ) + SetEvent( handle->condition ); + else { // spawn a thread to stop the stream + unsigned threadId; + stream_.callbackInfo.thread = _beginthreadex( NULL, 0, &asioStopStream, + &stream_.callbackInfo, 0, &threadId ); + } + return SUCCESS; + } + + // 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; + } + int cbReturnValue = callback( stream_.userBuffer[0], stream_.userBuffer[1], + stream_.bufferSize, streamTime, status, info->userData ); + if ( cbReturnValue == 2 ) { + stream_.state = STREAM_STOPPING; + handle->drainCounter = 2; + unsigned threadId; + stream_.callbackInfo.thread = _beginthreadex( NULL, 0, &asioStopStream, + &stream_.callbackInfo, 0, &threadId ); + return SUCCESS; + } + else if ( cbReturnValue == 1 ) { + 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 ( 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 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 { + + if ( stream_.doByteSwap[0] ) + byteSwapBuffer( stream_.userBuffer[0], + stream_.bufferSize * stream_.nUserChannels[0], + stream_.userFormat ); + + 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 ); + } + + } + } + + // Don't bother draining input + if ( handle->drainCounter ) { + handle->drainCounter++; + goto unlock; + } + + 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] ); + + } + 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(); + + RtApi::tickStreamTime(); + return SUCCESS; +} + +static 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 ( RtAudioError &exception ) { + std::cerr << "\nRtApiAsio: sampleRateChanged() error (" << exception.getMessage() << ")!\n" << std::endl; + return; + } + + std::cerr << "\nRtApiAsio: driver reports sample rate changed to " << sRate << " ... stream stopped!!!\n" << std::endl; +} + +static 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; + 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 ) +{ + struct Messages + { + ASIOError value; + const char*message; + }; + + static const 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; + + return "Unknown error."; +} + +//******************** End of __WINDOWS_ASIO__ *********************// +#endif + + +#if defined(__WINDOWS_WASAPI__) // Windows WASAPI API + +// Authored by Marcus Tomlinson <themarcustomlinson@gmail.com>, April 2014 +// - Introduces support for the Windows WASAPI API +// - Aims to deliver bit streams to and from hardware at the lowest possible latency, via the absolute minimum buffer sizes required +// - Provides flexible stream configuration to an otherwise strict and inflexible WASAPI interface +// - Includes automatic internal conversion of sample rate and buffer size between hardware and the user + +#ifndef INITGUID + #define INITGUID +#endif +#include <audioclient.h> +#include <avrt.h> +#include <mmdeviceapi.h> +#include <functiondiscoverykeys_devpkey.h> + +//============================================================================= + +#define SAFE_RELEASE( objectPtr )\ +if ( objectPtr )\ +{\ + objectPtr->Release();\ + objectPtr = NULL;\ +} + +typedef HANDLE ( __stdcall *TAvSetMmThreadCharacteristicsPtr )( LPCWSTR TaskName, LPDWORD TaskIndex ); + +//----------------------------------------------------------------------------- + +// WASAPI dictates stream sample rate, format, channel count, and in some cases, buffer size. +// Therefore we must perform all necessary conversions to user buffers in order to satisfy these +// requirements. WasapiBuffer ring buffers are used between HwIn->UserIn and UserOut->HwOut to +// provide intermediate storage for read / write synchronization. +class WasapiBuffer +{ +public: + WasapiBuffer() + : buffer_( NULL ), + bufferSize_( 0 ), + inIndex_( 0 ), + outIndex_( 0 ) {} + + ~WasapiBuffer() { + free( buffer_ ); + } + + // sets the length of the internal ring buffer + void setBufferSize( unsigned int bufferSize, unsigned int formatBytes ) { + free( buffer_ ); + + buffer_ = ( char* ) calloc( bufferSize, formatBytes ); + + bufferSize_ = bufferSize; + inIndex_ = 0; + outIndex_ = 0; + } + + // attempt to push a buffer into the ring buffer at the current "in" index + bool pushBuffer( char* buffer, unsigned int bufferSize, RtAudioFormat format ) + { + if ( !buffer || // incoming buffer is NULL + bufferSize == 0 || // incoming buffer has no data + bufferSize > bufferSize_ ) // incoming buffer too large + { + return false; + } + + unsigned int relOutIndex = outIndex_; + unsigned int inIndexEnd = inIndex_ + bufferSize; + if ( relOutIndex < inIndex_ && inIndexEnd >= bufferSize_ ) { + relOutIndex += bufferSize_; + } + + // "in" index can end on the "out" index but cannot begin at it + if ( inIndex_ <= relOutIndex && inIndexEnd > relOutIndex ) { + return false; // not enough space between "in" index and "out" index + } + + // copy buffer from external to internal + int fromZeroSize = inIndex_ + bufferSize - bufferSize_; + fromZeroSize = fromZeroSize < 0 ? 0 : fromZeroSize; + int fromInSize = bufferSize - fromZeroSize; + + switch( format ) + { + case RTAUDIO_SINT8: + memcpy( &( ( char* ) buffer_ )[inIndex_], buffer, fromInSize * sizeof( char ) ); + memcpy( buffer_, &( ( char* ) buffer )[fromInSize], fromZeroSize * sizeof( char ) ); + break; + case RTAUDIO_SINT16: + memcpy( &( ( short* ) buffer_ )[inIndex_], buffer, fromInSize * sizeof( short ) ); + memcpy( buffer_, &( ( short* ) buffer )[fromInSize], fromZeroSize * sizeof( short ) ); + break; + case RTAUDIO_SINT24: + memcpy( &( ( S24* ) buffer_ )[inIndex_], buffer, fromInSize * sizeof( S24 ) ); + memcpy( buffer_, &( ( S24* ) buffer )[fromInSize], fromZeroSize * sizeof( S24 ) ); + break; + case RTAUDIO_SINT32: + memcpy( &( ( int* ) buffer_ )[inIndex_], buffer, fromInSize * sizeof( int ) ); + memcpy( buffer_, &( ( int* ) buffer )[fromInSize], fromZeroSize * sizeof( int ) ); + break; + case RTAUDIO_FLOAT32: + memcpy( &( ( float* ) buffer_ )[inIndex_], buffer, fromInSize * sizeof( float ) ); + memcpy( buffer_, &( ( float* ) buffer )[fromInSize], fromZeroSize * sizeof( float ) ); + break; + case RTAUDIO_FLOAT64: + memcpy( &( ( double* ) buffer_ )[inIndex_], buffer, fromInSize * sizeof( double ) ); + memcpy( buffer_, &( ( double* ) buffer )[fromInSize], fromZeroSize * sizeof( double ) ); + break; + } + + // update "in" index + inIndex_ += bufferSize; + inIndex_ %= bufferSize_; + + return true; + } + + // attempt to pull a buffer from the ring buffer from the current "out" index + bool pullBuffer( char* buffer, unsigned int bufferSize, RtAudioFormat format ) + { + if ( !buffer || // incoming buffer is NULL + bufferSize == 0 || // incoming buffer has no data + bufferSize > bufferSize_ ) // incoming buffer too large + { + return false; + } + + unsigned int relInIndex = inIndex_; + unsigned int outIndexEnd = outIndex_ + bufferSize; + if ( relInIndex < outIndex_ && outIndexEnd >= bufferSize_ ) { + relInIndex += bufferSize_; + } + + // "out" index can begin at and end on the "in" index + if ( outIndex_ < relInIndex && outIndexEnd > relInIndex ) { + return false; // not enough space between "out" index and "in" index + } + + // copy buffer from internal to external + int fromZeroSize = outIndex_ + bufferSize - bufferSize_; + fromZeroSize = fromZeroSize < 0 ? 0 : fromZeroSize; + int fromOutSize = bufferSize - fromZeroSize; + + switch( format ) + { + case RTAUDIO_SINT8: + memcpy( buffer, &( ( char* ) buffer_ )[outIndex_], fromOutSize * sizeof( char ) ); + memcpy( &( ( char* ) buffer )[fromOutSize], buffer_, fromZeroSize * sizeof( char ) ); + break; + case RTAUDIO_SINT16: + memcpy( buffer, &( ( short* ) buffer_ )[outIndex_], fromOutSize * sizeof( short ) ); + memcpy( &( ( short* ) buffer )[fromOutSize], buffer_, fromZeroSize * sizeof( short ) ); + break; + case RTAUDIO_SINT24: + memcpy( buffer, &( ( S24* ) buffer_ )[outIndex_], fromOutSize * sizeof( S24 ) ); + memcpy( &( ( S24* ) buffer )[fromOutSize], buffer_, fromZeroSize * sizeof( S24 ) ); + break; + case RTAUDIO_SINT32: + memcpy( buffer, &( ( int* ) buffer_ )[outIndex_], fromOutSize * sizeof( int ) ); + memcpy( &( ( int* ) buffer )[fromOutSize], buffer_, fromZeroSize * sizeof( int ) ); + break; + case RTAUDIO_FLOAT32: + memcpy( buffer, &( ( float* ) buffer_ )[outIndex_], fromOutSize * sizeof( float ) ); + memcpy( &( ( float* ) buffer )[fromOutSize], buffer_, fromZeroSize * sizeof( float ) ); + break; + case RTAUDIO_FLOAT64: + memcpy( buffer, &( ( double* ) buffer_ )[outIndex_], fromOutSize * sizeof( double ) ); + memcpy( &( ( double* ) buffer )[fromOutSize], buffer_, fromZeroSize * sizeof( double ) ); + break; + } + + // update "out" index + outIndex_ += bufferSize; + outIndex_ %= bufferSize_; + + return true; + } + +private: + char* buffer_; + unsigned int bufferSize_; + unsigned int inIndex_; + unsigned int outIndex_; +}; + +//----------------------------------------------------------------------------- + +// In order to satisfy WASAPI's buffer requirements, we need a means of converting sample rate +// between HW and the user. The convertBufferWasapi function is used to perform this conversion +// between HwIn->UserIn and UserOut->HwOut during the stream callback loop. +// This sample rate converter favors speed over quality, and works best with conversions between +// one rate and its multiple. +void convertBufferWasapi( char* outBuffer, + const char* inBuffer, + const unsigned int& channelCount, + const unsigned int& inSampleRate, + const unsigned int& outSampleRate, + const unsigned int& inSampleCount, + unsigned int& outSampleCount, + const RtAudioFormat& format ) +{ + // calculate the new outSampleCount and relative sampleStep + float sampleRatio = ( float ) outSampleRate / inSampleRate; + float sampleStep = 1.0f / sampleRatio; + float inSampleFraction = 0.0f; + + outSampleCount = ( unsigned int ) roundf( inSampleCount * sampleRatio ); + + // frame-by-frame, copy each relative input sample into it's corresponding output sample + for ( unsigned int outSample = 0; outSample < outSampleCount; outSample++ ) + { + unsigned int inSample = ( unsigned int ) inSampleFraction; + + switch ( format ) + { + case RTAUDIO_SINT8: + memcpy( &( ( char* ) outBuffer )[ outSample * channelCount ], &( ( char* ) inBuffer )[ inSample * channelCount ], channelCount * sizeof( char ) ); + break; + case RTAUDIO_SINT16: + memcpy( &( ( short* ) outBuffer )[ outSample * channelCount ], &( ( short* ) inBuffer )[ inSample * channelCount ], channelCount * sizeof( short ) ); + break; + case RTAUDIO_SINT24: + memcpy( &( ( S24* ) outBuffer )[ outSample * channelCount ], &( ( S24* ) inBuffer )[ inSample * channelCount ], channelCount * sizeof( S24 ) ); + break; + case RTAUDIO_SINT32: + memcpy( &( ( int* ) outBuffer )[ outSample * channelCount ], &( ( int* ) inBuffer )[ inSample * channelCount ], channelCount * sizeof( int ) ); + break; + case RTAUDIO_FLOAT32: + memcpy( &( ( float* ) outBuffer )[ outSample * channelCount ], &( ( float* ) inBuffer )[ inSample * channelCount ], channelCount * sizeof( float ) ); + break; + case RTAUDIO_FLOAT64: + memcpy( &( ( double* ) outBuffer )[ outSample * channelCount ], &( ( double* ) inBuffer )[ inSample * channelCount ], channelCount * sizeof( double ) ); + break; + } + + // jump to next in sample + inSampleFraction += sampleStep; + } +} + +//----------------------------------------------------------------------------- + +// A structure to hold various information related to the WASAPI implementation. +struct WasapiHandle +{ + IAudioClient* captureAudioClient; + IAudioClient* renderAudioClient; + IAudioCaptureClient* captureClient; + IAudioRenderClient* renderClient; + HANDLE captureEvent; + HANDLE renderEvent; + + WasapiHandle() + : captureAudioClient( NULL ), + renderAudioClient( NULL ), + captureClient( NULL ), + renderClient( NULL ), + captureEvent( NULL ), + renderEvent( NULL ) {} +}; + +//============================================================================= + +RtApiWasapi::RtApiWasapi() + : coInitialized_( false ), deviceEnumerator_( NULL ) +{ + // WASAPI can run either apartment or multi-threaded + HRESULT hr = CoInitialize( NULL ); + if ( !FAILED( hr ) ) + coInitialized_ = true; + + // Instantiate device enumerator + hr = CoCreateInstance( __uuidof( MMDeviceEnumerator ), NULL, + CLSCTX_ALL, __uuidof( IMMDeviceEnumerator ), + ( void** ) &deviceEnumerator_ ); + + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::RtApiWasapi: Unable to instantiate device enumerator"; + error( RtAudioError::DRIVER_ERROR ); + } +} + +//----------------------------------------------------------------------------- + +RtApiWasapi::~RtApiWasapi() +{ + if ( stream_.state != STREAM_CLOSED ) + closeStream(); + + SAFE_RELEASE( deviceEnumerator_ ); + + // If this object previously called CoInitialize() + if ( coInitialized_ ) + CoUninitialize(); +} + +//============================================================================= + +unsigned int RtApiWasapi::getDeviceCount( void ) +{ + unsigned int captureDeviceCount = 0; + unsigned int renderDeviceCount = 0; + + IMMDeviceCollection* captureDevices = NULL; + IMMDeviceCollection* renderDevices = NULL; + + // Count capture devices + errorText_.clear(); + HRESULT hr = deviceEnumerator_->EnumAudioEndpoints( eCapture, DEVICE_STATE_ACTIVE, &captureDevices ); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::getDeviceCount: Unable to retrieve capture device collection."; + goto Exit; + } + + hr = captureDevices->GetCount( &captureDeviceCount ); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::getDeviceCount: Unable to retrieve capture device count."; + goto Exit; + } + + // Count render devices + hr = deviceEnumerator_->EnumAudioEndpoints( eRender, DEVICE_STATE_ACTIVE, &renderDevices ); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::getDeviceCount: Unable to retrieve render device collection."; + goto Exit; + } + + hr = renderDevices->GetCount( &renderDeviceCount ); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::getDeviceCount: Unable to retrieve render device count."; + goto Exit; + } + +Exit: + // release all references + SAFE_RELEASE( captureDevices ); + SAFE_RELEASE( renderDevices ); + + if ( errorText_.empty() ) + return captureDeviceCount + renderDeviceCount; + + error( RtAudioError::DRIVER_ERROR ); + return 0; +} + +//----------------------------------------------------------------------------- + +RtAudio::DeviceInfo RtApiWasapi::getDeviceInfo( unsigned int device ) +{ + RtAudio::DeviceInfo info; + unsigned int captureDeviceCount = 0; + unsigned int renderDeviceCount = 0; + std::string defaultDeviceName; + bool isCaptureDevice = false; + + PROPVARIANT deviceNameProp; + PROPVARIANT defaultDeviceNameProp; + + IMMDeviceCollection* captureDevices = NULL; + IMMDeviceCollection* renderDevices = NULL; + IMMDevice* devicePtr = NULL; + IMMDevice* defaultDevicePtr = NULL; + IAudioClient* audioClient = NULL; + IPropertyStore* devicePropStore = NULL; + IPropertyStore* defaultDevicePropStore = NULL; + + WAVEFORMATEX* deviceFormat = NULL; + WAVEFORMATEX* closestMatchFormat = NULL; + + // probed + info.probed = false; + + // Count capture devices + errorText_.clear(); + RtAudioError::Type errorType = RtAudioError::DRIVER_ERROR; + HRESULT hr = deviceEnumerator_->EnumAudioEndpoints( eCapture, DEVICE_STATE_ACTIVE, &captureDevices ); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve capture device collection."; + goto Exit; + } + + hr = captureDevices->GetCount( &captureDeviceCount ); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve capture device count."; + goto Exit; + } + + // Count render devices + hr = deviceEnumerator_->EnumAudioEndpoints( eRender, DEVICE_STATE_ACTIVE, &renderDevices ); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve render device collection."; + goto Exit; + } + + hr = renderDevices->GetCount( &renderDeviceCount ); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve render device count."; + goto Exit; + } + + // validate device index + if ( device >= captureDeviceCount + renderDeviceCount ) { + errorText_ = "RtApiWasapi::getDeviceInfo: Invalid device index."; + errorType = RtAudioError::INVALID_USE; + goto Exit; + } + + // determine whether index falls within capture or render devices + if ( device >= renderDeviceCount ) { + hr = captureDevices->Item( device - renderDeviceCount, &devicePtr ); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve capture device handle."; + goto Exit; + } + isCaptureDevice = true; + } + else { + hr = renderDevices->Item( device, &devicePtr ); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve render device handle."; + goto Exit; + } + isCaptureDevice = false; + } + + // get default device name + if ( isCaptureDevice ) { + hr = deviceEnumerator_->GetDefaultAudioEndpoint( eCapture, eConsole, &defaultDevicePtr ); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve default capture device handle."; + goto Exit; + } + } + else { + hr = deviceEnumerator_->GetDefaultAudioEndpoint( eRender, eConsole, &defaultDevicePtr ); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve default render device handle."; + goto Exit; + } + } + + hr = defaultDevicePtr->OpenPropertyStore( STGM_READ, &defaultDevicePropStore ); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::getDeviceInfo: Unable to open default device property store."; + goto Exit; + } + PropVariantInit( &defaultDeviceNameProp ); + + hr = defaultDevicePropStore->GetValue( PKEY_Device_FriendlyName, &defaultDeviceNameProp ); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve default device property: PKEY_Device_FriendlyName."; + goto Exit; + } + + defaultDeviceName = convertCharPointerToStdString(defaultDeviceNameProp.pwszVal); + + // name + hr = devicePtr->OpenPropertyStore( STGM_READ, &devicePropStore ); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::getDeviceInfo: Unable to open device property store."; + goto Exit; + } + + PropVariantInit( &deviceNameProp ); + + hr = devicePropStore->GetValue( PKEY_Device_FriendlyName, &deviceNameProp ); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve device property: PKEY_Device_FriendlyName."; + goto Exit; + } + + info.name =convertCharPointerToStdString(deviceNameProp.pwszVal); + + // is default + if ( isCaptureDevice ) { + info.isDefaultInput = info.name == defaultDeviceName; + info.isDefaultOutput = false; + } + else { + info.isDefaultInput = false; + info.isDefaultOutput = info.name == defaultDeviceName; + } + + // channel count + hr = devicePtr->Activate( __uuidof( IAudioClient ), CLSCTX_ALL, NULL, ( void** ) &audioClient ); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve device audio client."; + goto Exit; + } + + hr = audioClient->GetMixFormat( &deviceFormat ); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve device mix format."; + goto Exit; + } + + if ( isCaptureDevice ) { + info.inputChannels = deviceFormat->nChannels; + info.outputChannels = 0; + info.duplexChannels = 0; + } + else { + info.inputChannels = 0; + info.outputChannels = deviceFormat->nChannels; + info.duplexChannels = 0; + } + + // sample rates + info.sampleRates.clear(); + + // allow support for all sample rates as we have a built-in sample rate converter + for ( unsigned int i = 0; i < MAX_SAMPLE_RATES; i++ ) { + info.sampleRates.push_back( SAMPLE_RATES[i] ); + } + info.preferredSampleRate = deviceFormat->nSamplesPerSec; + + // native format + info.nativeFormats = 0; + + if ( deviceFormat->wFormatTag == WAVE_FORMAT_IEEE_FLOAT || + ( deviceFormat->wFormatTag == WAVE_FORMAT_EXTENSIBLE && + ( ( WAVEFORMATEXTENSIBLE* ) deviceFormat )->SubFormat == KSDATAFORMAT_SUBTYPE_IEEE_FLOAT ) ) + { + if ( deviceFormat->wBitsPerSample == 32 ) { + info.nativeFormats |= RTAUDIO_FLOAT32; + } + else if ( deviceFormat->wBitsPerSample == 64 ) { + info.nativeFormats |= RTAUDIO_FLOAT64; + } + } + else if ( deviceFormat->wFormatTag == WAVE_FORMAT_PCM || + ( deviceFormat->wFormatTag == WAVE_FORMAT_EXTENSIBLE && + ( ( WAVEFORMATEXTENSIBLE* ) deviceFormat )->SubFormat == KSDATAFORMAT_SUBTYPE_PCM ) ) + { + if ( deviceFormat->wBitsPerSample == 8 ) { + info.nativeFormats |= RTAUDIO_SINT8; + } + else if ( deviceFormat->wBitsPerSample == 16 ) { + info.nativeFormats |= RTAUDIO_SINT16; + } + else if ( deviceFormat->wBitsPerSample == 24 ) { + info.nativeFormats |= RTAUDIO_SINT24; + } + else if ( deviceFormat->wBitsPerSample == 32 ) { + info.nativeFormats |= RTAUDIO_SINT32; + } + } + + // probed + info.probed = true; + +Exit: + // release all references + PropVariantClear( &deviceNameProp ); + PropVariantClear( &defaultDeviceNameProp ); + + SAFE_RELEASE( captureDevices ); + SAFE_RELEASE( renderDevices ); + SAFE_RELEASE( devicePtr ); + SAFE_RELEASE( defaultDevicePtr ); + SAFE_RELEASE( audioClient ); + SAFE_RELEASE( devicePropStore ); + SAFE_RELEASE( defaultDevicePropStore ); + + CoTaskMemFree( deviceFormat ); + CoTaskMemFree( closestMatchFormat ); + + if ( !errorText_.empty() ) + error( errorType ); + return info; +} + +//----------------------------------------------------------------------------- + +unsigned int RtApiWasapi::getDefaultOutputDevice( void ) +{ + for ( unsigned int i = 0; i < getDeviceCount(); i++ ) { + if ( getDeviceInfo( i ).isDefaultOutput ) { + return i; + } + } + + return 0; +} + +//----------------------------------------------------------------------------- + +unsigned int RtApiWasapi::getDefaultInputDevice( void ) +{ + for ( unsigned int i = 0; i < getDeviceCount(); i++ ) { + if ( getDeviceInfo( i ).isDefaultInput ) { + return i; + } + } + + return 0; +} + +//----------------------------------------------------------------------------- + +void RtApiWasapi::closeStream( void ) +{ + if ( stream_.state == STREAM_CLOSED ) { + errorText_ = "RtApiWasapi::closeStream: No open stream to close."; + error( RtAudioError::WARNING ); + return; + } + + if ( stream_.state != STREAM_STOPPED ) + stopStream(); + + // clean up stream memory + SAFE_RELEASE( ( ( WasapiHandle* ) stream_.apiHandle )->captureAudioClient ) + SAFE_RELEASE( ( ( WasapiHandle* ) stream_.apiHandle )->renderAudioClient ) + + SAFE_RELEASE( ( ( WasapiHandle* ) stream_.apiHandle )->captureClient ) + SAFE_RELEASE( ( ( WasapiHandle* ) stream_.apiHandle )->renderClient ) + + if ( ( ( WasapiHandle* ) stream_.apiHandle )->captureEvent ) + CloseHandle( ( ( WasapiHandle* ) stream_.apiHandle )->captureEvent ); + + if ( ( ( WasapiHandle* ) stream_.apiHandle )->renderEvent ) + CloseHandle( ( ( WasapiHandle* ) stream_.apiHandle )->renderEvent ); + + delete ( WasapiHandle* ) stream_.apiHandle; + stream_.apiHandle = NULL; + + 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; + } + + // update stream state + stream_.state = STREAM_CLOSED; +} + +//----------------------------------------------------------------------------- + +void RtApiWasapi::startStream( void ) +{ + verifyStream(); + + if ( stream_.state == STREAM_RUNNING ) { + errorText_ = "RtApiWasapi::startStream: The stream is already running."; + error( RtAudioError::WARNING ); + return; + } + + // update stream state + stream_.state = STREAM_RUNNING; + + // create WASAPI stream thread + stream_.callbackInfo.thread = ( ThreadHandle ) CreateThread( NULL, 0, runWasapiThread, this, CREATE_SUSPENDED, NULL ); + + if ( !stream_.callbackInfo.thread ) { + errorText_ = "RtApiWasapi::startStream: Unable to instantiate callback thread."; + error( RtAudioError::THREAD_ERROR ); + } + else { + SetThreadPriority( ( void* ) stream_.callbackInfo.thread, stream_.callbackInfo.priority ); + ResumeThread( ( void* ) stream_.callbackInfo.thread ); + } +} + +//----------------------------------------------------------------------------- + +void RtApiWasapi::stopStream( void ) +{ + verifyStream(); + + if ( stream_.state == STREAM_STOPPED ) { + errorText_ = "RtApiWasapi::stopStream: The stream is already stopped."; + error( RtAudioError::WARNING ); + return; + } + + // inform stream thread by setting stream state to STREAM_STOPPING + stream_.state = STREAM_STOPPING; + + // wait until stream thread is stopped + while( stream_.state != STREAM_STOPPED ) { + Sleep( 1 ); + } + + // Wait for the last buffer to play before stopping. + Sleep( 1000 * stream_.bufferSize / stream_.sampleRate ); + + // stop capture client if applicable + if ( ( ( WasapiHandle* ) stream_.apiHandle )->captureAudioClient ) { + HRESULT hr = ( ( WasapiHandle* ) stream_.apiHandle )->captureAudioClient->Stop(); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::stopStream: Unable to stop capture stream."; + error( RtAudioError::DRIVER_ERROR ); + return; + } + } + + // stop render client if applicable + if ( ( ( WasapiHandle* ) stream_.apiHandle )->renderAudioClient ) { + HRESULT hr = ( ( WasapiHandle* ) stream_.apiHandle )->renderAudioClient->Stop(); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::stopStream: Unable to stop render stream."; + error( RtAudioError::DRIVER_ERROR ); + return; + } + } + + // close thread handle + if ( stream_.callbackInfo.thread && !CloseHandle( ( void* ) stream_.callbackInfo.thread ) ) { + errorText_ = "RtApiWasapi::stopStream: Unable to close callback thread."; + error( RtAudioError::THREAD_ERROR ); + return; + } + + stream_.callbackInfo.thread = (ThreadHandle) NULL; +} + +//----------------------------------------------------------------------------- + +void RtApiWasapi::abortStream( void ) +{ + verifyStream(); + + if ( stream_.state == STREAM_STOPPED ) { + errorText_ = "RtApiWasapi::abortStream: The stream is already stopped."; + error( RtAudioError::WARNING ); + return; + } + + // inform stream thread by setting stream state to STREAM_STOPPING + stream_.state = STREAM_STOPPING; + + // wait until stream thread is stopped + while ( stream_.state != STREAM_STOPPED ) { + Sleep( 1 ); + } + + // stop capture client if applicable + if ( ( ( WasapiHandle* ) stream_.apiHandle )->captureAudioClient ) { + HRESULT hr = ( ( WasapiHandle* ) stream_.apiHandle )->captureAudioClient->Stop(); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::abortStream: Unable to stop capture stream."; + error( RtAudioError::DRIVER_ERROR ); + return; + } + } + + // stop render client if applicable + if ( ( ( WasapiHandle* ) stream_.apiHandle )->renderAudioClient ) { + HRESULT hr = ( ( WasapiHandle* ) stream_.apiHandle )->renderAudioClient->Stop(); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::abortStream: Unable to stop render stream."; + error( RtAudioError::DRIVER_ERROR ); + return; + } + } + + // close thread handle + if ( stream_.callbackInfo.thread && !CloseHandle( ( void* ) stream_.callbackInfo.thread ) ) { + errorText_ = "RtApiWasapi::abortStream: Unable to close callback thread."; + error( RtAudioError::THREAD_ERROR ); + return; + } + + stream_.callbackInfo.thread = (ThreadHandle) NULL; +} + +//----------------------------------------------------------------------------- + +bool RtApiWasapi::probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels, + unsigned int firstChannel, unsigned int sampleRate, + RtAudioFormat format, unsigned int* bufferSize, + RtAudio::StreamOptions* options ) +{ + bool methodResult = FAILURE; + unsigned int captureDeviceCount = 0; + unsigned int renderDeviceCount = 0; + + IMMDeviceCollection* captureDevices = NULL; + IMMDeviceCollection* renderDevices = NULL; + IMMDevice* devicePtr = NULL; + WAVEFORMATEX* deviceFormat = NULL; + unsigned int bufferBytes; + stream_.state = STREAM_STOPPED; + + // create API Handle if not already created + if ( !stream_.apiHandle ) + stream_.apiHandle = ( void* ) new WasapiHandle(); + + // Count capture devices + errorText_.clear(); + RtAudioError::Type errorType = RtAudioError::DRIVER_ERROR; + HRESULT hr = deviceEnumerator_->EnumAudioEndpoints( eCapture, DEVICE_STATE_ACTIVE, &captureDevices ); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve capture device collection."; + goto Exit; + } + + hr = captureDevices->GetCount( &captureDeviceCount ); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve capture device count."; + goto Exit; + } + + // Count render devices + hr = deviceEnumerator_->EnumAudioEndpoints( eRender, DEVICE_STATE_ACTIVE, &renderDevices ); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve render device collection."; + goto Exit; + } + + hr = renderDevices->GetCount( &renderDeviceCount ); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve render device count."; + goto Exit; + } + + // validate device index + if ( device >= captureDeviceCount + renderDeviceCount ) { + errorType = RtAudioError::INVALID_USE; + errorText_ = "RtApiWasapi::probeDeviceOpen: Invalid device index."; + goto Exit; + } + + // determine whether index falls within capture or render devices + if ( device >= renderDeviceCount ) { + if ( mode != INPUT ) { + errorType = RtAudioError::INVALID_USE; + errorText_ = "RtApiWasapi::probeDeviceOpen: Capture device selected as output device."; + goto Exit; + } + + // retrieve captureAudioClient from devicePtr + IAudioClient*& captureAudioClient = ( ( WasapiHandle* ) stream_.apiHandle )->captureAudioClient; + + hr = captureDevices->Item( device - renderDeviceCount, &devicePtr ); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve capture device handle."; + goto Exit; + } + + hr = devicePtr->Activate( __uuidof( IAudioClient ), CLSCTX_ALL, + NULL, ( void** ) &captureAudioClient ); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve device audio client."; + goto Exit; + } + + hr = captureAudioClient->GetMixFormat( &deviceFormat ); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve device mix format."; + goto Exit; + } + + stream_.nDeviceChannels[mode] = deviceFormat->nChannels; + captureAudioClient->GetStreamLatency( ( long long* ) &stream_.latency[mode] ); + } + else { + if ( mode != OUTPUT ) { + errorType = RtAudioError::INVALID_USE; + errorText_ = "RtApiWasapi::probeDeviceOpen: Render device selected as input device."; + goto Exit; + } + + // retrieve renderAudioClient from devicePtr + IAudioClient*& renderAudioClient = ( ( WasapiHandle* ) stream_.apiHandle )->renderAudioClient; + + hr = renderDevices->Item( device, &devicePtr ); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve render device handle."; + goto Exit; + } + + hr = devicePtr->Activate( __uuidof( IAudioClient ), CLSCTX_ALL, + NULL, ( void** ) &renderAudioClient ); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve device audio client."; + goto Exit; + } + + hr = renderAudioClient->GetMixFormat( &deviceFormat ); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve device mix format."; + goto Exit; + } + + stream_.nDeviceChannels[mode] = deviceFormat->nChannels; + renderAudioClient->GetStreamLatency( ( long long* ) &stream_.latency[mode] ); + } + + // fill stream data + if ( ( stream_.mode == OUTPUT && mode == INPUT ) || + ( stream_.mode == INPUT && mode == OUTPUT ) ) { + stream_.mode = DUPLEX; + } + else { + stream_.mode = mode; + } + + stream_.device[mode] = device; + stream_.doByteSwap[mode] = false; + stream_.sampleRate = sampleRate; + stream_.bufferSize = *bufferSize; + stream_.nBuffers = 1; + stream_.nUserChannels[mode] = channels; + stream_.channelOffset[mode] = firstChannel; + stream_.userFormat = format; + stream_.deviceFormat[mode] = getDeviceInfo( device ).nativeFormats; + + if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) + stream_.userInterleaved = false; + else + stream_.userInterleaved = true; + stream_.deviceInterleaved[mode] = true; + + // Set flags for buffer conversion. + stream_.doConvertBuffer[mode] = false; + if ( stream_.userFormat != stream_.deviceFormat[mode] || + stream_.nUserChannels != stream_.nDeviceChannels ) + stream_.doConvertBuffer[mode] = true; + else if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] && + stream_.nUserChannels[mode] > 1 ) + stream_.doConvertBuffer[mode] = true; + + if ( stream_.doConvertBuffer[mode] ) + setConvertInfo( mode, 0 ); + + // Allocate necessary internal buffers + bufferBytes = stream_.nUserChannels[mode] * stream_.bufferSize * formatBytes( stream_.userFormat ); + + stream_.userBuffer[mode] = ( char* ) calloc( bufferBytes, 1 ); + if ( !stream_.userBuffer[mode] ) { + errorType = RtAudioError::MEMORY_ERROR; + errorText_ = "RtApiWasapi::probeDeviceOpen: Error allocating user buffer memory."; + goto Exit; + } + + if ( options && options->flags & RTAUDIO_SCHEDULE_REALTIME ) + stream_.callbackInfo.priority = 15; + else + stream_.callbackInfo.priority = 0; + + ///! TODO: RTAUDIO_MINIMIZE_LATENCY // Provide stream buffers directly to callback + ///! TODO: RTAUDIO_HOG_DEVICE // Exclusive mode + + methodResult = SUCCESS; + +Exit: + //clean up + SAFE_RELEASE( captureDevices ); + SAFE_RELEASE( renderDevices ); + SAFE_RELEASE( devicePtr ); + CoTaskMemFree( deviceFormat ); + + // if method failed, close the stream + if ( methodResult == FAILURE ) + closeStream(); + + if ( !errorText_.empty() ) + error( errorType ); + return methodResult; +} + +//============================================================================= + +DWORD WINAPI RtApiWasapi::runWasapiThread( void* wasapiPtr ) +{ + if ( wasapiPtr ) + ( ( RtApiWasapi* ) wasapiPtr )->wasapiThread(); + + return 0; +} + +DWORD WINAPI RtApiWasapi::stopWasapiThread( void* wasapiPtr ) +{ + if ( wasapiPtr ) + ( ( RtApiWasapi* ) wasapiPtr )->stopStream(); + + return 0; +} + +DWORD WINAPI RtApiWasapi::abortWasapiThread( void* wasapiPtr ) +{ + if ( wasapiPtr ) + ( ( RtApiWasapi* ) wasapiPtr )->abortStream(); + + return 0; +} + +//----------------------------------------------------------------------------- + +void RtApiWasapi::wasapiThread() +{ + // as this is a new thread, we must CoInitialize it + CoInitialize( NULL ); + + HRESULT hr; + + IAudioClient* captureAudioClient = ( ( WasapiHandle* ) stream_.apiHandle )->captureAudioClient; + IAudioClient* renderAudioClient = ( ( WasapiHandle* ) stream_.apiHandle )->renderAudioClient; + IAudioCaptureClient* captureClient = ( ( WasapiHandle* ) stream_.apiHandle )->captureClient; + IAudioRenderClient* renderClient = ( ( WasapiHandle* ) stream_.apiHandle )->renderClient; + HANDLE captureEvent = ( ( WasapiHandle* ) stream_.apiHandle )->captureEvent; + HANDLE renderEvent = ( ( WasapiHandle* ) stream_.apiHandle )->renderEvent; + + WAVEFORMATEX* captureFormat = NULL; + WAVEFORMATEX* renderFormat = NULL; + float captureSrRatio = 0.0f; + float renderSrRatio = 0.0f; + WasapiBuffer captureBuffer; + WasapiBuffer renderBuffer; + + // declare local stream variables + RtAudioCallback callback = ( RtAudioCallback ) stream_.callbackInfo.callback; + BYTE* streamBuffer = NULL; + unsigned long captureFlags = 0; + unsigned int bufferFrameCount = 0; + unsigned int numFramesPadding = 0; + unsigned int convBufferSize = 0; + bool callbackPushed = false; + bool callbackPulled = false; + bool callbackStopped = false; + int callbackResult = 0; + + // convBuffer is used to store converted buffers between WASAPI and the user + char* convBuffer = NULL; + unsigned int convBuffSize = 0; + unsigned int deviceBuffSize = 0; + + errorText_.clear(); + RtAudioError::Type errorType = RtAudioError::DRIVER_ERROR; + + // Attempt to assign "Pro Audio" characteristic to thread + HMODULE AvrtDll = LoadLibrary( (LPCTSTR) "AVRT.dll" ); + if ( AvrtDll ) { + DWORD taskIndex = 0; + TAvSetMmThreadCharacteristicsPtr AvSetMmThreadCharacteristicsPtr = ( TAvSetMmThreadCharacteristicsPtr ) GetProcAddress( AvrtDll, "AvSetMmThreadCharacteristicsW" ); + AvSetMmThreadCharacteristicsPtr( L"Pro Audio", &taskIndex ); + FreeLibrary( AvrtDll ); + } + + // start capture stream if applicable + if ( captureAudioClient ) { + hr = captureAudioClient->GetMixFormat( &captureFormat ); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::wasapiThread: Unable to retrieve device mix format."; + goto Exit; + } + + captureSrRatio = ( ( float ) captureFormat->nSamplesPerSec / stream_.sampleRate ); + + // initialize capture stream according to desire buffer size + float desiredBufferSize = stream_.bufferSize * captureSrRatio; + REFERENCE_TIME desiredBufferPeriod = ( REFERENCE_TIME ) ( ( float ) desiredBufferSize * 10000000 / captureFormat->nSamplesPerSec ); + + if ( !captureClient ) { + hr = captureAudioClient->Initialize( AUDCLNT_SHAREMODE_SHARED, + AUDCLNT_STREAMFLAGS_EVENTCALLBACK, + desiredBufferPeriod, + desiredBufferPeriod, + captureFormat, + NULL ); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::wasapiThread: Unable to initialize capture audio client."; + goto Exit; + } + + hr = captureAudioClient->GetService( __uuidof( IAudioCaptureClient ), + ( void** ) &captureClient ); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::wasapiThread: Unable to retrieve capture client handle."; + goto Exit; + } + + // configure captureEvent to trigger on every available capture buffer + captureEvent = CreateEvent( NULL, FALSE, FALSE, NULL ); + if ( !captureEvent ) { + errorType = RtAudioError::SYSTEM_ERROR; + errorText_ = "RtApiWasapi::wasapiThread: Unable to create capture event."; + goto Exit; + } + + hr = captureAudioClient->SetEventHandle( captureEvent ); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::wasapiThread: Unable to set capture event handle."; + goto Exit; + } + + ( ( WasapiHandle* ) stream_.apiHandle )->captureClient = captureClient; + ( ( WasapiHandle* ) stream_.apiHandle )->captureEvent = captureEvent; + } + + unsigned int inBufferSize = 0; + hr = captureAudioClient->GetBufferSize( &inBufferSize ); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::wasapiThread: Unable to get capture buffer size."; + goto Exit; + } + + // scale outBufferSize according to stream->user sample rate ratio + unsigned int outBufferSize = ( unsigned int ) ( stream_.bufferSize * captureSrRatio ) * stream_.nDeviceChannels[INPUT]; + inBufferSize *= stream_.nDeviceChannels[INPUT]; + + // set captureBuffer size + captureBuffer.setBufferSize( inBufferSize + outBufferSize, formatBytes( stream_.deviceFormat[INPUT] ) ); + + // reset the capture stream + hr = captureAudioClient->Reset(); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::wasapiThread: Unable to reset capture stream."; + goto Exit; + } + + // start the capture stream + hr = captureAudioClient->Start(); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::wasapiThread: Unable to start capture stream."; + goto Exit; + } + } + + // start render stream if applicable + if ( renderAudioClient ) { + hr = renderAudioClient->GetMixFormat( &renderFormat ); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::wasapiThread: Unable to retrieve device mix format."; + goto Exit; + } + + renderSrRatio = ( ( float ) renderFormat->nSamplesPerSec / stream_.sampleRate ); + + // initialize render stream according to desire buffer size + float desiredBufferSize = stream_.bufferSize * renderSrRatio; + REFERENCE_TIME desiredBufferPeriod = ( REFERENCE_TIME ) ( ( float ) desiredBufferSize * 10000000 / renderFormat->nSamplesPerSec ); + + if ( !renderClient ) { + hr = renderAudioClient->Initialize( AUDCLNT_SHAREMODE_SHARED, + AUDCLNT_STREAMFLAGS_EVENTCALLBACK, + desiredBufferPeriod, + desiredBufferPeriod, + renderFormat, + NULL ); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::wasapiThread: Unable to initialize render audio client."; + goto Exit; + } + + hr = renderAudioClient->GetService( __uuidof( IAudioRenderClient ), + ( void** ) &renderClient ); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::wasapiThread: Unable to retrieve render client handle."; + goto Exit; + } + + // configure renderEvent to trigger on every available render buffer + renderEvent = CreateEvent( NULL, FALSE, FALSE, NULL ); + if ( !renderEvent ) { + errorType = RtAudioError::SYSTEM_ERROR; + errorText_ = "RtApiWasapi::wasapiThread: Unable to create render event."; + goto Exit; + } + + hr = renderAudioClient->SetEventHandle( renderEvent ); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::wasapiThread: Unable to set render event handle."; + goto Exit; + } + + ( ( WasapiHandle* ) stream_.apiHandle )->renderClient = renderClient; + ( ( WasapiHandle* ) stream_.apiHandle )->renderEvent = renderEvent; + } + + unsigned int outBufferSize = 0; + hr = renderAudioClient->GetBufferSize( &outBufferSize ); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::wasapiThread: Unable to get render buffer size."; + goto Exit; + } + + // scale inBufferSize according to user->stream sample rate ratio + unsigned int inBufferSize = ( unsigned int ) ( stream_.bufferSize * renderSrRatio ) * stream_.nDeviceChannels[OUTPUT]; + outBufferSize *= stream_.nDeviceChannels[OUTPUT]; + + // set renderBuffer size + renderBuffer.setBufferSize( inBufferSize + outBufferSize, formatBytes( stream_.deviceFormat[OUTPUT] ) ); + + // reset the render stream + hr = renderAudioClient->Reset(); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::wasapiThread: Unable to reset render stream."; + goto Exit; + } + + // start the render stream + hr = renderAudioClient->Start(); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::wasapiThread: Unable to start render stream."; + goto Exit; + } + } + + if ( stream_.mode == INPUT ) { + convBuffSize = ( size_t ) ( stream_.bufferSize * captureSrRatio ) * stream_.nDeviceChannels[INPUT] * formatBytes( stream_.deviceFormat[INPUT] ); + deviceBuffSize = stream_.bufferSize * stream_.nDeviceChannels[INPUT] * formatBytes( stream_.deviceFormat[INPUT] ); + } + else if ( stream_.mode == OUTPUT ) { + convBuffSize = ( size_t ) ( stream_.bufferSize * renderSrRatio ) * stream_.nDeviceChannels[OUTPUT] * formatBytes( stream_.deviceFormat[OUTPUT] ); + deviceBuffSize = stream_.bufferSize * stream_.nDeviceChannels[OUTPUT] * formatBytes( stream_.deviceFormat[OUTPUT] ); + } + else if ( stream_.mode == DUPLEX ) { + convBuffSize = std::max( ( size_t ) ( stream_.bufferSize * captureSrRatio ) * stream_.nDeviceChannels[INPUT] * formatBytes( stream_.deviceFormat[INPUT] ), + ( size_t ) ( stream_.bufferSize * renderSrRatio ) * stream_.nDeviceChannels[OUTPUT] * formatBytes( stream_.deviceFormat[OUTPUT] ) ); + deviceBuffSize = std::max( stream_.bufferSize * stream_.nDeviceChannels[INPUT] * formatBytes( stream_.deviceFormat[INPUT] ), + stream_.bufferSize * stream_.nDeviceChannels[OUTPUT] * formatBytes( stream_.deviceFormat[OUTPUT] ) ); + } + + convBuffer = ( char* ) malloc( convBuffSize ); + stream_.deviceBuffer = ( char* ) malloc( deviceBuffSize ); + if ( !convBuffer || !stream_.deviceBuffer ) { + errorType = RtAudioError::MEMORY_ERROR; + errorText_ = "RtApiWasapi::wasapiThread: Error allocating device buffer memory."; + goto Exit; + } + + // stream process loop + while ( stream_.state != STREAM_STOPPING ) { + if ( !callbackPulled ) { + // Callback Input + // ============== + // 1. Pull callback buffer from inputBuffer + // 2. If 1. was successful: Convert callback buffer to user sample rate and channel count + // Convert callback buffer to user format + + if ( captureAudioClient ) { + // Pull callback buffer from inputBuffer + callbackPulled = captureBuffer.pullBuffer( convBuffer, + ( unsigned int ) ( stream_.bufferSize * captureSrRatio ) * stream_.nDeviceChannels[INPUT], + stream_.deviceFormat[INPUT] ); + + if ( callbackPulled ) { + // Convert callback buffer to user sample rate + convertBufferWasapi( stream_.deviceBuffer, + convBuffer, + stream_.nDeviceChannels[INPUT], + captureFormat->nSamplesPerSec, + stream_.sampleRate, + ( unsigned int ) ( stream_.bufferSize * captureSrRatio ), + convBufferSize, + stream_.deviceFormat[INPUT] ); + + if ( stream_.doConvertBuffer[INPUT] ) { + // Convert callback buffer to user format + convertBuffer( stream_.userBuffer[INPUT], + stream_.deviceBuffer, + stream_.convertInfo[INPUT] ); + } + else { + // no further conversion, simple copy deviceBuffer to userBuffer + memcpy( stream_.userBuffer[INPUT], + stream_.deviceBuffer, + stream_.bufferSize * stream_.nUserChannels[INPUT] * formatBytes( stream_.userFormat ) ); + } + } + } + else { + // if there is no capture stream, set callbackPulled flag + callbackPulled = true; + } + + // Execute Callback + // ================ + // 1. Execute user callback method + // 2. Handle return value from callback + + // if callback has not requested the stream to stop + if ( callbackPulled && !callbackStopped ) { + // Execute user callback method + callbackResult = callback( stream_.userBuffer[OUTPUT], + stream_.userBuffer[INPUT], + stream_.bufferSize, + getStreamTime(), + captureFlags & AUDCLNT_BUFFERFLAGS_DATA_DISCONTINUITY ? RTAUDIO_INPUT_OVERFLOW : 0, + stream_.callbackInfo.userData ); + + // Handle return value from callback + if ( callbackResult == 1 ) { + // instantiate a thread to stop this thread + HANDLE threadHandle = CreateThread( NULL, 0, stopWasapiThread, this, 0, NULL ); + if ( !threadHandle ) { + errorType = RtAudioError::THREAD_ERROR; + errorText_ = "RtApiWasapi::wasapiThread: Unable to instantiate stream stop thread."; + goto Exit; + } + else if ( !CloseHandle( threadHandle ) ) { + errorType = RtAudioError::THREAD_ERROR; + errorText_ = "RtApiWasapi::wasapiThread: Unable to close stream stop thread handle."; + goto Exit; + } + + callbackStopped = true; + } + else if ( callbackResult == 2 ) { + // instantiate a thread to stop this thread + HANDLE threadHandle = CreateThread( NULL, 0, abortWasapiThread, this, 0, NULL ); + if ( !threadHandle ) { + errorType = RtAudioError::THREAD_ERROR; + errorText_ = "RtApiWasapi::wasapiThread: Unable to instantiate stream abort thread."; + goto Exit; + } + else if ( !CloseHandle( threadHandle ) ) { + errorType = RtAudioError::THREAD_ERROR; + errorText_ = "RtApiWasapi::wasapiThread: Unable to close stream abort thread handle."; + goto Exit; + } + + callbackStopped = true; + } + } + } + + // Callback Output + // =============== + // 1. Convert callback buffer to stream format + // 2. Convert callback buffer to stream sample rate and channel count + // 3. Push callback buffer into outputBuffer + + if ( renderAudioClient && callbackPulled ) { + if ( stream_.doConvertBuffer[OUTPUT] ) { + // Convert callback buffer to stream format + convertBuffer( stream_.deviceBuffer, + stream_.userBuffer[OUTPUT], + stream_.convertInfo[OUTPUT] ); + + } + + // Convert callback buffer to stream sample rate + convertBufferWasapi( convBuffer, + stream_.deviceBuffer, + stream_.nDeviceChannels[OUTPUT], + stream_.sampleRate, + renderFormat->nSamplesPerSec, + stream_.bufferSize, + convBufferSize, + stream_.deviceFormat[OUTPUT] ); + + // Push callback buffer into outputBuffer + callbackPushed = renderBuffer.pushBuffer( convBuffer, + convBufferSize * stream_.nDeviceChannels[OUTPUT], + stream_.deviceFormat[OUTPUT] ); + } + else { + // if there is no render stream, set callbackPushed flag + callbackPushed = true; + } + + // Stream Capture + // ============== + // 1. Get capture buffer from stream + // 2. Push capture buffer into inputBuffer + // 3. If 2. was successful: Release capture buffer + + if ( captureAudioClient ) { + // if the callback input buffer was not pulled from captureBuffer, wait for next capture event + if ( !callbackPulled ) { + WaitForSingleObject( captureEvent, INFINITE ); + } + + // Get capture buffer from stream + hr = captureClient->GetBuffer( &streamBuffer, + &bufferFrameCount, + &captureFlags, NULL, NULL ); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::wasapiThread: Unable to retrieve capture buffer."; + goto Exit; + } + + if ( bufferFrameCount != 0 ) { + // Push capture buffer into inputBuffer + if ( captureBuffer.pushBuffer( ( char* ) streamBuffer, + bufferFrameCount * stream_.nDeviceChannels[INPUT], + stream_.deviceFormat[INPUT] ) ) + { + // Release capture buffer + hr = captureClient->ReleaseBuffer( bufferFrameCount ); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::wasapiThread: Unable to release capture buffer."; + goto Exit; + } + } + else + { + // Inform WASAPI that capture was unsuccessful + hr = captureClient->ReleaseBuffer( 0 ); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::wasapiThread: Unable to release capture buffer."; + goto Exit; + } + } + } + else + { + // Inform WASAPI that capture was unsuccessful + hr = captureClient->ReleaseBuffer( 0 ); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::wasapiThread: Unable to release capture buffer."; + goto Exit; + } + } + } + + // Stream Render + // ============= + // 1. Get render buffer from stream + // 2. Pull next buffer from outputBuffer + // 3. If 2. was successful: Fill render buffer with next buffer + // Release render buffer + + if ( renderAudioClient ) { + // if the callback output buffer was not pushed to renderBuffer, wait for next render event + if ( callbackPulled && !callbackPushed ) { + WaitForSingleObject( renderEvent, INFINITE ); + } + + // Get render buffer from stream + hr = renderAudioClient->GetBufferSize( &bufferFrameCount ); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::wasapiThread: Unable to retrieve render buffer size."; + goto Exit; + } + + hr = renderAudioClient->GetCurrentPadding( &numFramesPadding ); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::wasapiThread: Unable to retrieve render buffer padding."; + goto Exit; + } + + bufferFrameCount -= numFramesPadding; + + if ( bufferFrameCount != 0 ) { + hr = renderClient->GetBuffer( bufferFrameCount, &streamBuffer ); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::wasapiThread: Unable to retrieve render buffer."; + goto Exit; + } + + // Pull next buffer from outputBuffer + // Fill render buffer with next buffer + if ( renderBuffer.pullBuffer( ( char* ) streamBuffer, + bufferFrameCount * stream_.nDeviceChannels[OUTPUT], + stream_.deviceFormat[OUTPUT] ) ) + { + // Release render buffer + hr = renderClient->ReleaseBuffer( bufferFrameCount, 0 ); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::wasapiThread: Unable to release render buffer."; + goto Exit; + } + } + else + { + // Inform WASAPI that render was unsuccessful + hr = renderClient->ReleaseBuffer( 0, 0 ); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::wasapiThread: Unable to release render buffer."; + goto Exit; + } + } + } + else + { + // Inform WASAPI that render was unsuccessful + hr = renderClient->ReleaseBuffer( 0, 0 ); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::wasapiThread: Unable to release render buffer."; + goto Exit; + } + } + } + + // if the callback buffer was pushed renderBuffer reset callbackPulled flag + if ( callbackPushed ) { + callbackPulled = false; + } + + // tick stream time + RtApi::tickStreamTime(); + } + +Exit: + // clean up + CoTaskMemFree( captureFormat ); + CoTaskMemFree( renderFormat ); + + free ( convBuffer ); + + CoUninitialize(); + + // update stream state + stream_.state = STREAM_STOPPED; + + if ( errorText_.empty() ) + return; + else + error( errorType ); +} + +//******************** End of __WINDOWS_WASAPI__ *********************// +#endif + + +#if defined(__WINDOWS_DS__) // Windows DirectSound API + +// 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 +#define WAVE_FORMAT_96M08 0x00010000 /* 96 kHz, Mono, 8-bit */ +#define WAVE_FORMAT_96S08 0x00020000 /* 96 kHz, Stereo, 8-bit */ +#define WAVE_FORMAT_96M16 0x00040000 /* 96 kHz, Mono, 16-bit */ +#define WAVE_FORMAT_96S16 0x00080000 /* 96 kHz, Stereo, 16-bit */ +#endif + +#define MINIMUM_DEVICE_BUFFER_SIZE 32768 + +#ifdef _MSC_VER // if Microsoft Visual C++ +#pragma comment( lib, "winmm.lib" ) // then, auto-link winmm.lib. Otherwise, it has to be added manually. +#endif + +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; } +}; + +// Declarations for utility functions, callbacks, and structures +// specific to the DirectSound implementation. +static BOOL CALLBACK deviceQueryCallback( LPGUID lpguid, + LPCTSTR description, + LPCTSTR module, + LPVOID lpContext ); + +static const char* getErrorString( int code ); + +static unsigned __stdcall callbackHandler( void *ptr ); + +struct DsDevice { + LPGUID id[2]; + bool validId[2]; + bool found; + std::string name; + + DsDevice() + : found(false) { validId[0] = false; validId[1] = false; } +}; + +struct DsProbeData { + bool isInput; + std::vector<struct DsDevice>* dsDevices; +}; + +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; +} + +RtApiDs :: ~RtApiDs() +{ + if ( coInitialized_ ) CoUninitialize(); // balanced call. + if ( stream_.state != STREAM_CLOSED ) closeStream(); +} + +// The DirectSound default output is always the first device. +unsigned int RtApiDs :: getDefaultOutputDevice( void ) +{ + return 0; +} + +// 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. + struct DsProbeData probeInfo; + probeInfo.isInput = false; + probeInfo.dsDevices = &dsDevices; + HRESULT result = DirectSoundEnumerate( (LPDSENUMCALLBACK) deviceQueryCallback, &probeInfo ); + if ( FAILED( result ) ) { + errorStream_ << "RtApiDs::getDeviceCount: error (" << getErrorString( result ) << ") enumerating output devices!"; + errorText_ = errorStream_.str(); + error( RtAudioError::WARNING ); + } + + // Query DirectSoundCapture devices. + probeInfo.isInput = true; + result = DirectSoundCaptureEnumerate( (LPDSENUMCALLBACK) deviceQueryCallback, &probeInfo ); + if ( FAILED( result ) ) { + errorStream_ << "RtApiDs::getDeviceCount: error (" << getErrorString( result ) << ") enumerating input devices!"; + errorText_ = errorStream_.str(); + error( RtAudioError::WARNING ); + } + + // Clean out any devices that may have disappeared (code update submitted by Eli Zehngut). + for ( unsigned int i=0; i<dsDevices.size(); ) { + if ( dsDevices[i].found == false ) dsDevices.erase( dsDevices.begin() + i ); + else i++; + } + + return static_cast<unsigned int>(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( RtAudioError::INVALID_USE ); + return info; + } + } + + if ( device >= dsDevices.size() ) { + errorText_ = "RtApiDs::getDeviceInfo: device ID is invalid!"; + error( RtAudioError::INVALID_USE ); + return info; + } + + HRESULT result; + if ( dsDevices[ device ].validId[0] == false ) goto probeInput; + + 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( RtAudioError::WARNING ); + goto probeInput; + } + + 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( RtAudioError::WARNING ); + goto probeInput; + } + + // 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] ); + + if ( !info.preferredSampleRate || ( SAMPLE_RATES[k] <= 48000 && SAMPLE_RATES[k] > info.preferredSampleRate ) ) + info.preferredSampleRate = SAMPLE_RATES[k]; + } + } + + // 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( RtAudioError::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( RtAudioError::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] ); + } + std::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; + } + + size_t 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(); + 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; + } + } + + // 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; + result = DirectSoundCreate( dsDevices[ device ].id[0], &output, NULL ); + if ( FAILED( result ) ) { + errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") opening output device (" << dsDevices[ device ].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 (" << dsDevices[ device ].name << ")!"; + errorText_ = errorStream_.str(); + return FAILURE; + } + + // 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; + } + + // 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; + + // 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; + + // 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 (" << dsDevices[ device ].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 (" << dsDevices[ device ].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 (" << dsDevices[ device ].name << ")!"; + errorText_ = errorStream_.str(); + return FAILURE; + } + + // 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; + } + } + + // 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; + } + + dsBufferSize = dsbcaps.dwBufferBytes; + + // 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; + } + + // Zero the DS buffer + ZeroMemory( audioPtr, dataLen ); + + // 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; + } + + 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; + } + + DSCCAPS inCaps; + inCaps.dwSize = sizeof( inCaps ); + result = input->GetCaps( &inCaps ); + if ( FAILED( result ) ) { + input->Release(); + errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") getting input capabilities (" << dsDevices[ device ].name << ")!"; + 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; + } + + // 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; + 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. + 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; + } + + dsBufferSize = dscbcaps.dwBufferBytes; + + // 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. + + // 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; + } + + // 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 (" << dsDevices[ device ].name << ")!"; + errorText_ = errorStream_.str(); + return FAILURE; + } + + 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 + 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; + } + + 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; + + 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 callback thread. + if ( stream_.callbackInfo.isRunning == false ) { + unsigned threadId; + stream_.callbackInfo.isRunning = true; + stream_.callbackInfo.object = (void *) this; + 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; + + 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; + } + + 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; + } + + stream_.state = STREAM_CLOSED; + return FAILURE; +} + +void RtApiDs :: closeStream() +{ + if ( stream_.state == STREAM_CLOSED ) { + errorText_ = "RtApiDs::closeStream(): no open stream to close!"; + error( RtAudioError::WARNING ); + return; + } + + // 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(); + } + CloseHandle( 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; + } + } + + if ( stream_.deviceBuffer ) { + free( stream_.deviceBuffer ); + stream_.deviceBuffer = 0; + } + + stream_.mode = UNINITIALIZED; + stream_.state = STREAM_CLOSED; +} + +void RtApiDs :: startStream() +{ + verifyStream(); + if ( stream_.state == STREAM_RUNNING ) { + errorText_ = "RtApiDs::startStream(): the stream is already running!"; + error( RtAudioError::WARNING ); + return; + } + + DsHandle *handle = (DsHandle *) stream_.apiHandle; + + // 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 ); + + 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] ); + } + + 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; + } + } + + 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; + } + } + + handle->drainCounter = 0; + handle->internalDrain = false; + ResetEvent( handle->condition ); + stream_.state = STREAM_RUNNING; + + unlock: + if ( FAILED( result ) ) error( RtAudioError::SYSTEM_ERROR ); +} + +void RtApiDs :: stopStream() +{ + verifyStream(); + if ( stream_.state == STREAM_STOPPED ) { + errorText_ = "RtApiDs::stopStream(): the stream is already stopped!"; + error( RtAudioError::WARNING ); + return; + } + + 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 = 2; + WaitForSingleObject( handle->condition, INFINITE ); // block until signaled + } + + stream_.state = STREAM_STOPPED; + + MUTEX_LOCK( &stream_.mutex ); + + // 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; + } + + // 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; + } + + // Zero the DS buffer + ZeroMemory( audioPtr, dataLen ); + + // 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; + } + + // If we start playing again, we must begin at beginning of buffer. + handle->bufferPointer[0] = 0; + } + + if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) { + LPDIRECTSOUNDCAPTUREBUFFER buffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1]; + audioPtr = NULL; + dataLen = 0; + + stream_.state = STREAM_STOPPED; + + if ( stream_.mode != DUPLEX ) + MUTEX_LOCK( &stream_.mutex ); + + result = buffer->Stop(); + if ( FAILED( result ) ) { + errorStream_ << "RtApiDs::stopStream: 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::stopStream: 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::stopStream: error (" << getErrorString( result ) << ") unlocking input buffer!"; + errorText_ = errorStream_.str(); + goto unlock; + } + + // If we start recording again, we must begin at beginning of buffer. + handle->bufferPointer[1] = 0; + } + + unlock: + timeEndPeriod( 1 ); // revert to normal scheduler frequency on lesser windows. + MUTEX_UNLOCK( &stream_.mutex ); + + if ( FAILED( result ) ) error( RtAudioError::SYSTEM_ERROR ); +} + +void RtApiDs :: abortStream() +{ + verifyStream(); + if ( stream_.state == STREAM_STOPPED ) { + errorText_ = "RtApiDs::abortStream(): the stream is already stopped!"; + error( RtAudioError::WARNING ); + return; + } + + DsHandle *handle = (DsHandle *) stream_.apiHandle; + handle->drainCounter = 2; + + stopStream(); +} + +void RtApiDs :: callbackEvent() +{ + if ( stream_.state == STREAM_STOPPED || stream_.state == STREAM_STOPPING ) { + Sleep( 50 ); // sleep 50 milliseconds + return; + } + + if ( stream_.state == STREAM_CLOSED ) { + errorText_ = "RtApiDs::callbackEvent(): the stream is closed ... this shouldn't happen!"; + error( RtAudioError::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 ) { + + stream_.state = STREAM_STOPPING; + if ( handle->internalDrain == false ) + SetEvent( handle->condition ); + else + stopStream(); + 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; + } + int cbReturnValue = callback( stream_.userBuffer[0], stream_.userBuffer[1], + stream_.bufferSize, streamTime, status, info->userData ); + if ( cbReturnValue == 2 ) { + stream_.state = STREAM_STOPPING; + handle->drainCounter = 2; + abortStream(); + return; + } + else if ( cbReturnValue == 1 ) { + 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; + + MUTEX_LOCK( &stream_.mutex ); + if ( stream_.state == STREAM_STOPPED ) { + MUTEX_UNLOCK( &stream_.mutex ); + return; + } + + 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 + // within a ms or so of correct. + + LPDIRECTSOUNDBUFFER dsWriteBuffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0]; + LPDIRECTSOUNDCAPTUREBUFFER dsCaptureBuffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1]; + + DWORD startSafeWritePointer, startSafeReadPointer; + + result = dsWriteBuffer->GetCurrentPosition( NULL, &startSafeWritePointer ); + if ( FAILED( result ) ) { + errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current write position!"; + errorText_ = errorStream_.str(); + MUTEX_UNLOCK( &stream_.mutex ); + error( RtAudioError::SYSTEM_ERROR ); + return; + } + result = dsCaptureBuffer->GetCurrentPosition( NULL, &startSafeReadPointer ); + if ( FAILED( result ) ) { + errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current read position!"; + errorText_ = errorStream_.str(); + MUTEX_UNLOCK( &stream_.mutex ); + error( RtAudioError::SYSTEM_ERROR ); + return; + } + while ( true ) { + result = dsWriteBuffer->GetCurrentPosition( NULL, &safeWritePointer ); + if ( FAILED( result ) ) { + errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current write position!"; + errorText_ = errorStream_.str(); + MUTEX_UNLOCK( &stream_.mutex ); + error( RtAudioError::SYSTEM_ERROR ); + return; + } + result = dsCaptureBuffer->GetCurrentPosition( NULL, &safeReadPointer ); + if ( FAILED( result ) ) { + errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current read position!"; + errorText_ = errorStream_.str(); + MUTEX_UNLOCK( &stream_.mutex ); + error( RtAudioError::SYSTEM_ERROR ); + return; + } + if ( safeWritePointer != startSafeWritePointer && safeReadPointer != startSafeReadPointer ) break; + Sleep( 1 ); + } + + //assert( handle->dsBufferSize[0] == handle->dsBufferSize[1] ); + + 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 ) { + + // 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(); + MUTEX_UNLOCK( &stream_.mutex ); + error( RtAudioError::SYSTEM_ERROR ); + return; + } + handle->bufferPointer[0] = safeWritePointer + handle->dsPointerLeadTime[0]; + if ( handle->bufferPointer[0] >= handle->dsBufferSize[0] ) handle->bufferPointer[0] -= handle->dsBufferSize[0]; + } + + buffersRolling = true; + } + + 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 ); + } + + // No byte swapping necessary in DirectSound implementation. + + // 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 ); + + DWORD dsBufferSize = handle->dsBufferSize[0]; + nextWritePointer = handle->bufferPointer[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 ) << ") getting current write position!"; + errorText_ = errorStream_.str(); + error( RtAudioError::SYSTEM_ERROR ); + return; + } + + // 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; + + // Check whether the entire write region is behind the play pointer. + if ( leadPointer >= endWrite ) break; + + // 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 ( 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; + } + + // 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(); + MUTEX_UNLOCK( &stream_.mutex ); + error( RtAudioError::SYSTEM_ERROR ); + return; + } + + // Copy our buffer into the DS buffer + CopyMemory( buffer1, buffer, bufferSize1 ); + if ( buffer2 != NULL ) CopyMemory( buffer2, buffer+bufferSize1, bufferSize2 ); + + // 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(); + MUTEX_UNLOCK( &stream_.mutex ); + error( RtAudioError::SYSTEM_ERROR ); + return; + } + nextWritePointer = ( nextWritePointer + bufferSize1 + bufferSize2 ) % dsBufferSize; + handle->bufferPointer[0] = nextWritePointer; + } + + // Don't bother draining input + if ( handle->drainCounter ) { + handle->drainCounter++; + goto unlock; + } + + 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(); + MUTEX_UNLOCK( &stream_.mutex ); + error( RtAudioError::SYSTEM_ERROR ); + return; + } + + 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 ( safeReadPointer < endRead && stream_.callbackInfo.isRunning ) { + // 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(); + MUTEX_UNLOCK( &stream_.mutex ); + error( RtAudioError::SYSTEM_ERROR ); + return; + } + + if ( safeReadPointer < (DWORD)nextReadPointer ) safeReadPointer += dsBufferSize; // unwrap offset + } + } + + // 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(); + MUTEX_UNLOCK( &stream_.mutex ); + error( RtAudioError::SYSTEM_ERROR ); + return; + } + + 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(); + MUTEX_UNLOCK( &stream_.mutex ); + error( RtAudioError::SYSTEM_ERROR ); + return; + } + handle->bufferPointer[1] = nextReadPointer; + + // 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] ); + } + + unlock: + MUTEX_UNLOCK( &stream_.mutex ); + RtApi::tickStreamTime(); +} + +// Definitions for utility functions and callbacks +// specific to the DirectSound implementation. + +static 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; +} + +static BOOL CALLBACK deviceQueryCallback( LPGUID lpguid, + LPCTSTR description, + LPCTSTR /*module*/, + LPVOID lpContext ) +{ + struct DsProbeData& probeInfo = *(struct DsProbeData*) lpContext; + std::vector<struct DsDevice>& dsDevices = *probeInfo.dsDevices; + + HRESULT hr; + bool validDevice = false; + if ( probeInfo.isInput == true ) { + DSCCAPS caps; + LPDIRECTSOUNDCAPTURE object; + + 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 ) + validDevice = true; + } + 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 = convertCharPointerToStdString( description ); + //if ( name == "Primary Sound Driver" || name == "Primary Sound Capture Driver" ) + if ( lpguid == NULL ) + name = "Default Device"; + if ( validDevice ) { + for ( unsigned int i=0; i<dsDevices.size(); i++ ) { + if ( dsDevices[i].name == name ) { + dsDevices[i].found = true; + if ( probeInfo.isInput ) { + dsDevices[i].id[1] = lpguid; + dsDevices[i].validId[1] = true; + } + else { + dsDevices[i].id[0] = lpguid; + dsDevices[i].validId[0] = true; + } + return TRUE; + } + } + + DsDevice device; + device.name = name; + device.found = true; + if ( probeInfo.isInput ) { + device.id[1] = lpguid; + device.validId[1] = true; + } + else { + device.id[0] = lpguid; + device.validId[0] = true; + } + dsDevices.push_back( device ); + } + + return TRUE; +} + +static const char* getErrorString( int code ) +{ + switch ( code ) { + + case DSERR_ALLOCATED: + return "Already allocated"; + + case DSERR_CONTROLUNAVAIL: + return "Control unavailable"; + + case DSERR_INVALIDPARAM: + return "Invalid parameter"; + + case DSERR_INVALIDCALL: + return "Invalid call"; + + case DSERR_GENERIC: + return "Generic error"; + + case DSERR_PRIOLEVELNEEDED: + return "Priority level needed"; + + case DSERR_OUTOFMEMORY: + return "Out of memory"; + + case DSERR_BADFORMAT: + return "The sample rate or the channel format is not supported"; + + case DSERR_UNSUPPORTED: + return "Not supported"; + + case DSERR_NODRIVER: + return "No driver"; + + case DSERR_ALREADYINITIALIZED: + return "Already initialized"; + + case DSERR_NOAGGREGATION: + return "No aggregation"; + + case DSERR_BUFFERLOST: + return "Buffer lost"; + + case DSERR_OTHERAPPHASPRIO: + return "Another application already has priority"; + + case DSERR_UNINITIALIZED: + return "Uninitialized"; + + default: + return "DirectSound unknown error"; + } +} +//******************** End of __WINDOWS_DS__ *********************// +#endif + + +#if defined(__LINUX_ALSA__) + +#include <alsa/asoundlib.h> +#include <unistd.h> + + // 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_cv; + bool runnable; + + AlsaHandle() + :synchronized(false), runnable(false) { xrun[0] = false; xrun[1] = false; } +}; + +static void *alsaCallbackHandler( void * ptr ); + +RtApiAlsa :: RtApiAlsa() +{ + // Nothing to do here. +} + +RtApiAlsa :: ~RtApiAlsa() +{ + if ( stream_.state != STREAM_CLOSED ) closeStream(); +} + +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( RtAudioError::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( RtAudioError::WARNING ); + break; + } + if ( subdevice < 0 ) + break; + nDevices++; + } + nextcard: + snd_ctl_close( handle ); + snd_card_next( &card ); + } + + result = snd_ctl_open( &handle, "default", 0 ); + if (result == 0) { + nDevices++; + snd_ctl_close( handle ); + } + + return nDevices; +} + +RtAudio::DeviceInfo RtApiAlsa :: getDeviceInfo( unsigned int device ) +{ + RtAudio::DeviceInfo info; + info.probed = false; + + unsigned nDevices = 0; + int result, subdevice, card; + char name[64]; + snd_ctl_t *chandle; + + // Count cards and devices + card = -1; + subdevice = -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( RtAudioError::WARNING ); + goto nextcard; + } + 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( RtAudioError::WARNING ); + break; + } + if ( subdevice < 0 ) break; + if ( nDevices == device ) { + sprintf( name, "hw:%d,%d", card, subdevice ); + goto foundDevice; + } + nDevices++; + } + nextcard: + snd_ctl_close( chandle ); + snd_card_next( &card ); + } + + result = snd_ctl_open( &chandle, "default", SND_CTL_NONBLOCK ); + if ( result == 0 ) { + if ( nDevices == device ) { + strcpy( name, "default" ); + goto foundDevice; + } + nDevices++; + } + + if ( nDevices == 0 ) { + errorText_ = "RtApiAlsa::getDeviceInfo: no devices found!"; + error( RtAudioError::INVALID_USE ); + return info; + } + + if ( device >= nDevices ) { + errorText_ = "RtApiAlsa::getDeviceInfo: device ID is invalid!"; + error( RtAudioError::INVALID_USE ); + return info; + } + + 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 ) ) { + snd_ctl_close( chandle ); + if ( device >= devices_.size() ) { + errorText_ = "RtApiAlsa::getDeviceInfo: device ID was not present before stream was opened."; + error( RtAudioError::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 ); + + // First try for playback unless default device (which has subdev -1) + stream = SND_PCM_STREAM_PLAYBACK; + snd_pcm_info_set_stream( pcminfo, stream ); + if ( subdevice != -1 ) { + snd_pcm_info_set_device( pcminfo, subdevice ); + snd_pcm_info_set_subdevice( pcminfo, 0 ); + + 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( RtAudioError::WARNING ); + 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( RtAudioError::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( RtAudioError::WARNING ); + goto captureProbe; + } + info.outputChannels = value; + snd_pcm_close( phandle ); + + captureProbe: + stream = SND_PCM_STREAM_CAPTURE; + snd_pcm_info_set_stream( pcminfo, stream ); + + // Now try for capture unless default device (with subdev = -1) + if ( subdevice != -1 ) { + 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; + } + } + else + snd_ctl_close( chandle ); + + 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( RtAudioError::WARNING ); + 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( RtAudioError::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( RtAudioError::WARNING ); + if ( info.outputChannels == 0 ) return info; + goto probeParameters; + } + info.inputChannels = value; + snd_pcm_close( phandle ); + + // 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; + + // 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 ); + + 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( RtAudioError::WARNING ); + return info; + } + + // 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( RtAudioError::WARNING ); + return info; + } + + // 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.preferredSampleRate || ( SAMPLE_RATES[i] <= 48000 && SAMPLE_RATES[i] > info.preferredSampleRate ) ) + info.preferredSampleRate = 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( RtAudioError::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 ) { + snd_pcm_close( phandle ); + errorStream_ << "RtApiAlsa::getDeviceInfo: pcm device (" << name << ") data format not supported by RtAudio."; + errorText_ = errorStream_.str(); + error( RtAudioError::WARNING ); + return info; + } + + // Get the device name + char *cardname; + result = snd_card_get_name( card, &cardname ); + if ( result >= 0 ) { + sprintf( name, "hw:%s,%d", cardname, subdevice ); + free( cardname ); + } + 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); +#endif + + // 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; + + if ( options && options->flags & RTAUDIO_ALSA_USE_DEFAULT ) + snprintf(name, sizeof(name), "%s", "default"); + else { + // 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++; + } + snd_ctl_close( chandle ); + snd_card_next( &card ); + } + + result = snd_ctl_open( &chandle, "default", SND_CTL_NONBLOCK ); + if ( result == 0 ) { + if ( nDevices == device ) { + strcpy( name, "default" ); + goto foundDevice; + } + nDevices++; + } + + 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; + } + } + + 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(); + + 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 ) + 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; + } + + // 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 ); +#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; + 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; + } + + if ( result < 0 ) { + snd_pcm_close( phandle ); + errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting pcm device (" << name << ") access, " << snd_strerror( result ) << "."; + errorText_ = errorStream_.str(); + return FAILURE; + } + + // 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; + } + + // 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_FLOAT; + if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat ) == 0 ) { + stream_.deviceFormat[mode] = RTAUDIO_FLOAT32; + 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; + } + + 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. + snd_pcm_close( phandle ); + 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; + } + + // 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; + } + + // 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; + + // 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 (or period) size. + int dir = 0; + 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 number, which in ALSA is referred to as the "period". + unsigned int periods = 0; + if ( options && options->flags & RTAUDIO_MINIMIZE_LATENCY ) periods = 2; + if ( options && options->numberOfBuffers > 0 ) periods = options->numberOfBuffers; + if ( periods < 2 ) periods = 4; // a fairly safe default value + 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 ) { + snd_pcm_close( phandle ); + errorStream_ << "RtApiAlsa::probeDeviceOpen: system error setting buffer size for duplex stream on device (" << name << ")."; + errorText_ = errorStream_.str(); + return FAILURE; + } + + 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; + } + +#if defined(__RTAUDIO_DEBUG__) + 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; + } + +#if defined(__RTAUDIO_DEBUG__) + 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_cv, NULL ) ) { + errorText_ = "RtApiAlsa::probeDeviceOpen: error initializing pthread condition variable."; + goto error; + } + + stream_.apiHandle = (void *) apiInfo; + apiInfo->handles[0] = 0; + apiInfo->handles[1] = 0; + } + else { + apiInfo = (AlsaHandle *) stream_.apiHandle; + } + apiInfo->handles[mode] = phandle; + phandle = 0; + + // 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; + } + + 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_.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( RtAudioError::WARNING ); + } + } + 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 ) { + // We previously attempted to increase the audio callback priority + // to SCHED_RR here via the attributes. However, while no errors + // were reported in doing so, it did not work. So, now this is + // done in the alsaCallbackHandler function. + stream_.callbackInfo.doRealtime = true; + 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; + stream_.callbackInfo.priority = priority; + } +#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!"; + goto error; + } + } + + return SUCCESS; + + error: + if ( apiInfo ) { + pthread_cond_destroy( &apiInfo->runnable_cv ); + 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 ( phandle) snd_pcm_close( phandle ); + + 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; + } + + stream_.state = STREAM_CLOSED; + return FAILURE; +} + +void RtApiAlsa :: closeStream() +{ + if ( stream_.state == STREAM_CLOSED ) { + errorText_ = "RtApiAlsa::closeStream(): no open stream to close!"; + error( RtAudioError::WARNING ); + return; + } + + AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle; + stream_.callbackInfo.isRunning = false; + MUTEX_LOCK( &stream_.mutex ); + if ( stream_.state == STREAM_STOPPED ) { + apiInfo->runnable = true; + pthread_cond_signal( &apiInfo->runnable_cv ); + } + MUTEX_UNLOCK( &stream_.mutex ); + pthread_join( stream_.callbackInfo.thread, NULL ); + + 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 ( apiInfo ) { + pthread_cond_destroy( &apiInfo->runnable_cv ); + 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; + } + + stream_.mode = UNINITIALIZED; + stream_.state = STREAM_CLOSED; +} + +void RtApiAlsa :: startStream() +{ + // This method calls snd_pcm_prepare if the device isn't already in that state. + + verifyStream(); + if ( stream_.state == STREAM_RUNNING ) { + errorText_ = "RtApiAlsa::startStream(): the stream is already running!"; + error( RtAudioError::WARNING ); + return; + } + + MUTEX_LOCK( &stream_.mutex ); + + 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_.mode == INPUT || stream_.mode == DUPLEX ) && !apiInfo->synchronized ) { + result = snd_pcm_drop(handle[1]); // fix to remove stale data received since device has been open + 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; + } + } + } + + stream_.state = STREAM_RUNNING; + + unlock: + apiInfo->runnable = true; + pthread_cond_signal( &apiInfo->runnable_cv ); + MUTEX_UNLOCK( &stream_.mutex ); + + if ( result >= 0 ) return; + error( RtAudioError::SYSTEM_ERROR ); +} + +void RtApiAlsa :: stopStream() +{ + verifyStream(); + if ( stream_.state == STREAM_STOPPED ) { + errorText_ = "RtApiAlsa::stopStream(): the stream is already stopped!"; + error( RtAudioError::WARNING ); + return; + } + + stream_.state = STREAM_STOPPED; + 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] ); + 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::stopStream: error stopping input pcm device, " << snd_strerror( result ) << "."; + errorText_ = errorStream_.str(); + goto unlock; + } + } + + unlock: + apiInfo->runnable = false; // fixes high CPU usage when stopped + MUTEX_UNLOCK( &stream_.mutex ); + + if ( result >= 0 ) return; + error( RtAudioError::SYSTEM_ERROR ); +} + +void RtApiAlsa :: abortStream() +{ + verifyStream(); + if ( stream_.state == STREAM_STOPPED ) { + errorText_ = "RtApiAlsa::abortStream(): the stream is already stopped!"; + error( RtAudioError::WARNING ); + return; + } + + stream_.state = STREAM_STOPPED; + 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 ) { + 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 == 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; + } + } + + unlock: + apiInfo->runnable = false; // fixes high CPU usage when stopped + MUTEX_UNLOCK( &stream_.mutex ); + + if ( result >= 0 ) return; + error( RtAudioError::SYSTEM_ERROR ); +} + +void RtApiAlsa :: callbackEvent() +{ + AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle; + if ( stream_.state == STREAM_STOPPED ) { + MUTEX_LOCK( &stream_.mutex ); + while ( !apiInfo->runnable ) + pthread_cond_wait( &apiInfo->runnable_cv, &stream_.mutex ); + + if ( stream_.state != STREAM_RUNNING ) { + MUTEX_UNLOCK( &stream_.mutex ); + return; + } + MUTEX_UNLOCK( &stream_.mutex ); + } + + if ( stream_.state == STREAM_CLOSED ) { + errorText_ = "RtApiAlsa::callbackEvent(): the stream is closed ... this shouldn't happen!"; + error( RtAudioError::WARNING ); + 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; + } + 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 ); + + if ( doStopStream == 2 ) { + abortStream(); + return; + } + + MUTEX_LOCK( &stream_.mutex ); + + // The state might change while waiting on a mutex. + if ( stream_.state == STREAM_STOPPED ) goto unlock; + + int result; + char *buffer; + int channels; + snd_pcm_t **handle; + snd_pcm_sframes_t frames; + RtAudioFormat format; + handle = (snd_pcm_t **) apiInfo->handles; + + 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: error, current state is " << snd_pcm_state_name( state ) << ", " << snd_strerror( result ) << "."; + errorText_ = errorStream_.str(); + } + } + else { + errorStream_ << "RtApiAlsa::callbackEvent: audio read error, " << snd_strerror( result ) << "."; + errorText_ = errorStream_.str(); + } + error( RtAudioError::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] ); + + // Check stream latency + result = snd_pcm_delay( handle[1], &frames ); + if ( result == 0 && frames > 0 ) stream_.latency[1] = frames; + } + + tryOutput: + + if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { + + // 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; + } + + // Do byte swapping if necessary. + if ( stream_.doByteSwap[0] ) + byteSwapBuffer(buffer, stream_.bufferSize * channels, format); + + // 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 + errorText_ = "RtApiAlsa::callbackEvent: audio write error, underrun."; + } + else { + errorStream_ << "RtApiAlsa::callbackEvent: error, current state is " << snd_pcm_state_name( state ) << ", " << snd_strerror( result ) << "."; + errorText_ = errorStream_.str(); + } + } + else { + errorStream_ << "RtApiAlsa::callbackEvent: audio write error, " << snd_strerror( result ) << "."; + errorText_ = errorStream_.str(); + } + error( RtAudioError::WARNING ); + goto unlock; + } + + // Check stream latency + result = snd_pcm_delay( handle[0], &frames ); + if ( result == 0 && frames > 0 ) stream_.latency[0] = frames; + } + + unlock: + MUTEX_UNLOCK( &stream_.mutex ); + + RtApi::tickStreamTime(); + if ( doStopStream == 1 ) this->stopStream(); +} + +static void *alsaCallbackHandler( void *ptr ) +{ + CallbackInfo *info = (CallbackInfo *) ptr; + RtApiAlsa *object = (RtApiAlsa *) info->object; + bool *isRunning = &info->isRunning; + +#ifdef SCHED_RR // Undefined with some OSes (eg: NetBSD 1.6.x with GNU Pthread) + if ( &info->doRealtime ) { + pthread_t tID = pthread_self(); // ID of this thread + sched_param prio = { info->priority }; // scheduling priority of thread + pthread_setschedparam( tID, SCHED_RR, &prio ); + } +#endif + + while ( *isRunning == true ) { + pthread_testcancel(); + object->callbackEvent(); + } + + pthread_exit( NULL ); +} + +//******************** End of __LINUX_ALSA__ *********************// +#endif + +#if defined(__LINUX_PULSE__) + +// Code written by Peter Meerwald, pmeerw@pmeerw.net +// and Tristan Matthews. + +#include <pulse/error.h> +#include <pulse/simple.h> +#include <cstdio> + +static const unsigned int SUPPORTED_SAMPLERATES[] = { 8000, 16000, 22050, 32000, + 44100, 48000, 96000, 0}; + +struct rtaudio_pa_format_mapping_t { + RtAudioFormat rtaudio_format; + pa_sample_format_t pa_format; +}; + +static const rtaudio_pa_format_mapping_t supported_sampleformats[] = { + {RTAUDIO_SINT16, PA_SAMPLE_S16LE}, + {RTAUDIO_SINT32, PA_SAMPLE_S32LE}, + {RTAUDIO_FLOAT32, PA_SAMPLE_FLOAT32LE}, + {0, PA_SAMPLE_INVALID}}; + +struct PulseAudioHandle { + pa_simple *s_play; + pa_simple *s_rec; + pthread_t thread; + pthread_cond_t runnable_cv; + bool runnable; + PulseAudioHandle() : s_play(0), s_rec(0), runnable(false) { } +}; + +RtApiPulse::~RtApiPulse() +{ + if ( stream_.state != STREAM_CLOSED ) + closeStream(); +} + +unsigned int RtApiPulse::getDeviceCount( void ) +{ + return 1; +} + +RtAudio::DeviceInfo RtApiPulse::getDeviceInfo( unsigned int /*device*/ ) +{ + RtAudio::DeviceInfo info; + info.probed = true; + info.name = "PulseAudio"; + info.outputChannels = 2; + info.inputChannels = 2; + info.duplexChannels = 2; + info.isDefaultOutput = true; + info.isDefaultInput = true; + + for ( const unsigned int *sr = SUPPORTED_SAMPLERATES; *sr; ++sr ) + info.sampleRates.push_back( *sr ); + + info.preferredSampleRate = 48000; + info.nativeFormats = RTAUDIO_SINT16 | RTAUDIO_SINT32 | RTAUDIO_FLOAT32; + + return info; +} + +static void *pulseaudio_callback( void * user ) +{ + CallbackInfo *cbi = static_cast<CallbackInfo *>( user ); + RtApiPulse *context = static_cast<RtApiPulse *>( cbi->object ); + volatile bool *isRunning = &cbi->isRunning; + + while ( *isRunning ) { + pthread_testcancel(); + context->callbackEvent(); + } + + pthread_exit( NULL ); +} + +void RtApiPulse::closeStream( void ) +{ + PulseAudioHandle *pah = static_cast<PulseAudioHandle *>( stream_.apiHandle ); + + stream_.callbackInfo.isRunning = false; + if ( pah ) { + MUTEX_LOCK( &stream_.mutex ); + if ( stream_.state == STREAM_STOPPED ) { + pah->runnable = true; + pthread_cond_signal( &pah->runnable_cv ); + } + MUTEX_UNLOCK( &stream_.mutex ); + + pthread_join( pah->thread, 0 ); + if ( pah->s_play ) { + pa_simple_flush( pah->s_play, NULL ); + pa_simple_free( pah->s_play ); + } + if ( pah->s_rec ) + pa_simple_free( pah->s_rec ); + + pthread_cond_destroy( &pah->runnable_cv ); + delete pah; + stream_.apiHandle = 0; + } + + if ( stream_.userBuffer[0] ) { + free( stream_.userBuffer[0] ); + stream_.userBuffer[0] = 0; + } + if ( stream_.userBuffer[1] ) { + free( stream_.userBuffer[1] ); + stream_.userBuffer[1] = 0; + } + + stream_.state = STREAM_CLOSED; + stream_.mode = UNINITIALIZED; +} + +void RtApiPulse::callbackEvent( void ) +{ + PulseAudioHandle *pah = static_cast<PulseAudioHandle *>( stream_.apiHandle ); + + if ( stream_.state == STREAM_STOPPED ) { + MUTEX_LOCK( &stream_.mutex ); + while ( !pah->runnable ) + pthread_cond_wait( &pah->runnable_cv, &stream_.mutex ); + + if ( stream_.state != STREAM_RUNNING ) { + MUTEX_UNLOCK( &stream_.mutex ); + return; + } + MUTEX_UNLOCK( &stream_.mutex ); + } + + if ( stream_.state == STREAM_CLOSED ) { + errorText_ = "RtApiPulse::callbackEvent(): the stream is closed ... " + "this shouldn't happen!"; + error( RtAudioError::WARNING ); + return; + } + + RtAudioCallback callback = (RtAudioCallback) stream_.callbackInfo.callback; + double streamTime = getStreamTime(); + RtAudioStreamStatus status = 0; + int doStopStream = callback( stream_.userBuffer[OUTPUT], stream_.userBuffer[INPUT], + stream_.bufferSize, streamTime, status, + stream_.callbackInfo.userData ); + + if ( doStopStream == 2 ) { + abortStream(); + return; + } + + MUTEX_LOCK( &stream_.mutex ); + void *pulse_in = stream_.doConvertBuffer[INPUT] ? stream_.deviceBuffer : stream_.userBuffer[INPUT]; + void *pulse_out = stream_.doConvertBuffer[OUTPUT] ? stream_.deviceBuffer : stream_.userBuffer[OUTPUT]; + + if ( stream_.state != STREAM_RUNNING ) + goto unlock; + + int pa_error; + size_t bytes; + if (stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { + if ( stream_.doConvertBuffer[OUTPUT] ) { + convertBuffer( stream_.deviceBuffer, + stream_.userBuffer[OUTPUT], + stream_.convertInfo[OUTPUT] ); + bytes = stream_.nDeviceChannels[OUTPUT] * stream_.bufferSize * + formatBytes( stream_.deviceFormat[OUTPUT] ); + } else + bytes = stream_.nUserChannels[OUTPUT] * stream_.bufferSize * + formatBytes( stream_.userFormat ); + + if ( pa_simple_write( pah->s_play, pulse_out, bytes, &pa_error ) < 0 ) { + errorStream_ << "RtApiPulse::callbackEvent: audio write error, " << + pa_strerror( pa_error ) << "."; + errorText_ = errorStream_.str(); + error( RtAudioError::WARNING ); + } + } + + if ( stream_.mode == INPUT || stream_.mode == DUPLEX) { + if ( stream_.doConvertBuffer[INPUT] ) + bytes = stream_.nDeviceChannels[INPUT] * stream_.bufferSize * + formatBytes( stream_.deviceFormat[INPUT] ); + else + bytes = stream_.nUserChannels[INPUT] * stream_.bufferSize * + formatBytes( stream_.userFormat ); + + if ( pa_simple_read( pah->s_rec, pulse_in, bytes, &pa_error ) < 0 ) { + errorStream_ << "RtApiPulse::callbackEvent: audio read error, " << + pa_strerror( pa_error ) << "."; + errorText_ = errorStream_.str(); + error( RtAudioError::WARNING ); + } + if ( stream_.doConvertBuffer[INPUT] ) { + convertBuffer( stream_.userBuffer[INPUT], + stream_.deviceBuffer, + stream_.convertInfo[INPUT] ); + } + } + + unlock: + MUTEX_UNLOCK( &stream_.mutex ); + RtApi::tickStreamTime(); + + if ( doStopStream == 1 ) + stopStream(); +} + +void RtApiPulse::startStream( void ) +{ + PulseAudioHandle *pah = static_cast<PulseAudioHandle *>( stream_.apiHandle ); + + if ( stream_.state == STREAM_CLOSED ) { + errorText_ = "RtApiPulse::startStream(): the stream is not open!"; + error( RtAudioError::INVALID_USE ); + return; + } + if ( stream_.state == STREAM_RUNNING ) { + errorText_ = "RtApiPulse::startStream(): the stream is already running!"; + error( RtAudioError::WARNING ); + return; + } + + MUTEX_LOCK( &stream_.mutex ); + + stream_.state = STREAM_RUNNING; + + pah->runnable = true; + pthread_cond_signal( &pah->runnable_cv ); + MUTEX_UNLOCK( &stream_.mutex ); +} + +void RtApiPulse::stopStream( void ) +{ + PulseAudioHandle *pah = static_cast<PulseAudioHandle *>( stream_.apiHandle ); + + if ( stream_.state == STREAM_CLOSED ) { + errorText_ = "RtApiPulse::stopStream(): the stream is not open!"; + error( RtAudioError::INVALID_USE ); + return; + } + if ( stream_.state == STREAM_STOPPED ) { + errorText_ = "RtApiPulse::stopStream(): the stream is already stopped!"; + error( RtAudioError::WARNING ); + return; + } + + stream_.state = STREAM_STOPPED; + MUTEX_LOCK( &stream_.mutex ); + + if ( pah && pah->s_play ) { + int pa_error; + if ( pa_simple_drain( pah->s_play, &pa_error ) < 0 ) { + errorStream_ << "RtApiPulse::stopStream: error draining output device, " << + pa_strerror( pa_error ) << "."; + errorText_ = errorStream_.str(); + MUTEX_UNLOCK( &stream_.mutex ); + error( RtAudioError::SYSTEM_ERROR ); + return; + } + } + + stream_.state = STREAM_STOPPED; + MUTEX_UNLOCK( &stream_.mutex ); +} + +void RtApiPulse::abortStream( void ) +{ + PulseAudioHandle *pah = static_cast<PulseAudioHandle*>( stream_.apiHandle ); + + if ( stream_.state == STREAM_CLOSED ) { + errorText_ = "RtApiPulse::abortStream(): the stream is not open!"; + error( RtAudioError::INVALID_USE ); + return; + } + if ( stream_.state == STREAM_STOPPED ) { + errorText_ = "RtApiPulse::abortStream(): the stream is already stopped!"; + error( RtAudioError::WARNING ); + return; + } + + stream_.state = STREAM_STOPPED; + MUTEX_LOCK( &stream_.mutex ); + + if ( pah && pah->s_play ) { + int pa_error; + if ( pa_simple_flush( pah->s_play, &pa_error ) < 0 ) { + errorStream_ << "RtApiPulse::abortStream: error flushing output device, " << + pa_strerror( pa_error ) << "."; + errorText_ = errorStream_.str(); + MUTEX_UNLOCK( &stream_.mutex ); + error( RtAudioError::SYSTEM_ERROR ); + return; + } + } + + stream_.state = STREAM_STOPPED; + MUTEX_UNLOCK( &stream_.mutex ); +} + +bool RtApiPulse::probeDeviceOpen( unsigned int device, StreamMode mode, + unsigned int channels, unsigned int firstChannel, + unsigned int sampleRate, RtAudioFormat format, + unsigned int *bufferSize, RtAudio::StreamOptions *options ) +{ + PulseAudioHandle *pah = 0; + unsigned long bufferBytes = 0; + pa_sample_spec ss; + + if ( device != 0 ) return false; + if ( mode != INPUT && mode != OUTPUT ) return false; + if ( channels != 1 && channels != 2 ) { + errorText_ = "RtApiPulse::probeDeviceOpen: unsupported number of channels."; + return false; + } + ss.channels = channels; + + if ( firstChannel != 0 ) return false; + + bool sr_found = false; + for ( const unsigned int *sr = SUPPORTED_SAMPLERATES; *sr; ++sr ) { + if ( sampleRate == *sr ) { + sr_found = true; + stream_.sampleRate = sampleRate; + ss.rate = sampleRate; + break; + } + } + if ( !sr_found ) { + errorText_ = "RtApiPulse::probeDeviceOpen: unsupported sample rate."; + return false; + } + + bool sf_found = 0; + for ( const rtaudio_pa_format_mapping_t *sf = supported_sampleformats; + sf->rtaudio_format && sf->pa_format != PA_SAMPLE_INVALID; ++sf ) { + if ( format == sf->rtaudio_format ) { + sf_found = true; + stream_.userFormat = sf->rtaudio_format; + stream_.deviceFormat[mode] = stream_.userFormat; + ss.format = sf->pa_format; + break; + } + } + if ( !sf_found ) { // Use internal data format conversion. + stream_.userFormat = format; + stream_.deviceFormat[mode] = RTAUDIO_FLOAT32; + ss.format = PA_SAMPLE_FLOAT32LE; + } + + // Set other stream parameters. + if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) stream_.userInterleaved = false; + else stream_.userInterleaved = true; + stream_.deviceInterleaved[mode] = true; + stream_.nBuffers = 1; + stream_.doByteSwap[mode] = false; + stream_.nUserChannels[mode] = channels; + stream_.nDeviceChannels[mode] = channels + firstChannel; + stream_.channelOffset[mode] = 0; + std::string streamName = "RtAudio"; + + // 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; + + // 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_ = "RtApiPulse::probeDeviceOpen: error allocating user buffer memory."; + goto error; + } + stream_.bufferSize = *bufferSize; + + 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_ = "RtApiPulse::probeDeviceOpen: error allocating device buffer memory."; + goto error; + } + } + } + + stream_.device[mode] = device; + + // Setup the buffer conversion information structure. + if ( stream_.doConvertBuffer[mode] ) setConvertInfo( mode, firstChannel ); + + if ( !stream_.apiHandle ) { + PulseAudioHandle *pah = new PulseAudioHandle; + if ( !pah ) { + errorText_ = "RtApiPulse::probeDeviceOpen: error allocating memory for handle."; + goto error; + } + + stream_.apiHandle = pah; + if ( pthread_cond_init( &pah->runnable_cv, NULL ) != 0 ) { + errorText_ = "RtApiPulse::probeDeviceOpen: error creating condition variable."; + goto error; + } + } + pah = static_cast<PulseAudioHandle *>( stream_.apiHandle ); + + int error; + if ( options && !options->streamName.empty() ) streamName = options->streamName; + switch ( mode ) { + case INPUT: + pa_buffer_attr buffer_attr; + buffer_attr.fragsize = bufferBytes; + buffer_attr.maxlength = -1; + + pah->s_rec = pa_simple_new( NULL, streamName.c_str(), PA_STREAM_RECORD, NULL, "Record", &ss, NULL, &buffer_attr, &error ); + if ( !pah->s_rec ) { + errorText_ = "RtApiPulse::probeDeviceOpen: error connecting input to PulseAudio server."; + goto error; + } + break; + case OUTPUT: + pah->s_play = pa_simple_new( NULL, "RtAudio", PA_STREAM_PLAYBACK, NULL, "Playback", &ss, NULL, NULL, &error ); + if ( !pah->s_play ) { + errorText_ = "RtApiPulse::probeDeviceOpen: error connecting output to PulseAudio server."; + goto error; + } + break; + default: + goto error; + } + + if ( stream_.mode == UNINITIALIZED ) + stream_.mode = mode; + else if ( stream_.mode == mode ) + goto error; + else + stream_.mode = DUPLEX; + + if ( !stream_.callbackInfo.isRunning ) { + stream_.callbackInfo.object = this; + stream_.callbackInfo.isRunning = true; + if ( pthread_create( &pah->thread, NULL, pulseaudio_callback, (void *)&stream_.callbackInfo) != 0 ) { + errorText_ = "RtApiPulse::probeDeviceOpen: error creating thread."; + goto error; + } + } + + stream_.state = STREAM_STOPPED; + return true; + + error: + if ( pah && stream_.callbackInfo.isRunning ) { + pthread_cond_destroy( &pah->runnable_cv ); + delete pah; + 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; +} + +//******************** End of __LINUX_PULSE__ *********************// +#endif + +#if defined(__LINUX_OSS__) + +#include <unistd.h> +#include <sys/ioctl.h> +#include <unistd.h> +#include <fcntl.h> +#include <sys/soundcard.h> +#include <errno.h> +#include <math.h> + +static 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; } +}; + +RtApiOss :: RtApiOss() +{ + // Nothing to do here. +} + +RtApiOss :: ~RtApiOss() +{ + if ( stream_.state != STREAM_CLOSED ) closeStream(); +} + +unsigned int RtApiOss :: getDeviceCount( void ) +{ + int mixerfd = open( "/dev/mixer", O_RDWR, 0 ); + if ( mixerfd == -1 ) { + errorText_ = "RtApiOss::getDeviceCount: error opening '/dev/mixer'."; + error( RtAudioError::WARNING ); + return 0; + } + + 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( RtAudioError::WARNING ); + return 0; + } + + close( mixerfd ); + return sysinfo.numaudios; +} + +RtAudio::DeviceInfo RtApiOss :: getDeviceInfo( unsigned int device ) +{ + RtAudio::DeviceInfo info; + info.probed = false; + + int mixerfd = open( "/dev/mixer", O_RDWR, 0 ); + if ( mixerfd == -1 ) { + errorText_ = "RtApiOss::getDeviceInfo: error opening '/dev/mixer'."; + error( RtAudioError::WARNING ); + return info; + } + + 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( RtAudioError::WARNING ); + return info; + } + + unsigned nDevices = sysinfo.numaudios; + if ( nDevices == 0 ) { + close( mixerfd ); + errorText_ = "RtApiOss::getDeviceInfo: no devices found!"; + error( RtAudioError::INVALID_USE ); + return info; + } + + if ( device >= nDevices ) { + close( mixerfd ); + errorText_ = "RtApiOss::getDeviceInfo: device ID is invalid!"; + error( RtAudioError::INVALID_USE ); + 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( RtAudioError::WARNING ); + return info; + } + + // 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( RtAudioError::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] ); + + if ( !info.preferredSampleRate || ( SAMPLE_RATES[k] <= 48000 && SAMPLE_RATES[k] > info.preferredSampleRate ) ) + info.preferredSampleRate = SAMPLE_RATES[k]; + + break; + } + } + } + } + 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] ); + + if ( !info.preferredSampleRate || ( SAMPLE_RATES[k] <= 48000 && SAMPLE_RATES[k] > info.preferredSampleRate ) ) + info.preferredSampleRate = SAMPLE_RATES[k]; + } + } + } + + if ( info.sampleRates.size() == 0 ) { + errorStream_ << "RtApiOss::getDeviceInfo: no supported sample rates found for device (" << ainfo.name << ")."; + errorText_ = errorStream_.str(); + error( RtAudioError::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; + } + + 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; + } + + 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; + } + + 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; + } + + // 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; + } + + 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; + } + // 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; + + // 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; + } + } + */ + + // 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; + + // 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; + } + } + 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 ( 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_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; + } + } + + 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; + } + } + + 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 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; + } + + // 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; + + // Save buffer size (in sample frames). + *bufferSize = ossBufferBytes / ( formatBytes(stream_.deviceFormat[mode]) * deviceChannels ); + stream_.bufferSize = *bufferSize; + + // 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; + } + + // 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; + + 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; + } + + // 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; + + // 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 ( pthread_cond_init( &handle->runnable, NULL ) ) { + errorText_ = "RtApiOss::probeDeviceOpen: error initializing pthread condition variable."; + goto error; + } + + stream_.apiHandle = (void *) handle; + } + else { + handle = (OssHandle *) stream_.apiHandle; + } + 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."; + 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; + } + } + } + + 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; + if ( stream_.device[0] == device ) handle->id[0] = fd; + } + else { + stream_.mode = mode; + + // 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 ); +#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, 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; + } + + 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 RtApiOss :: closeStream() +{ + if ( stream_.state == STREAM_CLOSED ) { + errorText_ = "RtApiOss::closeStream(): no open stream to close!"; + error( RtAudioError::WARNING ); + return; + } + + 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 ( 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; + } + + 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; + } + + 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; + } + + stream_.mode = UNINITIALIZED; + stream_.state = STREAM_CLOSED; +} + +void RtApiOss :: startStream() +{ + verifyStream(); + if ( stream_.state == STREAM_RUNNING ) { + errorText_ = "RtApiOss::startStream(): the stream is already running!"; + error( RtAudioError::WARNING ); + return; + } + + MUTEX_LOCK( &stream_.mutex ); + + stream_.state = STREAM_RUNNING; + + // No need to do anything else here ... OSS automatically starts + // when fed samples. + + MUTEX_UNLOCK( &stream_.mutex ); + + OssHandle *handle = (OssHandle *) stream_.apiHandle; + pthread_cond_signal( &handle->runnable ); +} + +void RtApiOss :: stopStream() +{ + verifyStream(); + if ( stream_.state == STREAM_STOPPED ) { + errorText_ = "RtApiOss::stopStream(): the stream is already stopped!"; + error( RtAudioError::WARNING ); + return; + } + + MUTEX_LOCK( &stream_.mutex ); + + // The state might change while waiting on a mutex. + if ( stream_.state == STREAM_STOPPED ) { + MUTEX_UNLOCK( &stream_.mutex ); + return; + } + + int result = 0; + OssHandle *handle = (OssHandle *) stream_.apiHandle; + if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { + + // Flush the output with zeros a few times. + char *buffer; + int samples; + RtAudioFormat format; + + 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; + } + + 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( RtAudioError::WARNING ); + } + } + + 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_.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; + } + } + + unlock: + stream_.state = STREAM_STOPPED; + MUTEX_UNLOCK( &stream_.mutex ); + + if ( result != -1 ) return; + error( RtAudioError::SYSTEM_ERROR ); +} + +void RtApiOss :: abortStream() +{ + verifyStream(); + if ( stream_.state == STREAM_STOPPED ) { + errorText_ = "RtApiOss::abortStream(): the stream is already stopped!"; + error( RtAudioError::WARNING ); + return; + } + + MUTEX_LOCK( &stream_.mutex ); + + // The state might change while waiting on a mutex. + if ( stream_.state == STREAM_STOPPED ) { + MUTEX_UNLOCK( &stream_.mutex ); + return; + } + + 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_.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; + } + } + + unlock: + stream_.state = STREAM_STOPPED; + MUTEX_UNLOCK( &stream_.mutex ); + + if ( result != -1 ) return; + error( RtAudioError::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 ); + } + + if ( stream_.state == STREAM_CLOSED ) { + errorText_ = "RtApiOss::callbackEvent(): the stream is closed ... this shouldn't happen!"; + error( RtAudioError::WARNING ); + return; + } + + // 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; + } + + MUTEX_LOCK( &stream_.mutex ); + + // The state might change while waiting on a mutex. + if ( stream_.state == STREAM_STOPPED ) goto unlock; + + int result; + char *buffer; + int samples; + RtAudioFormat format; + + if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { + + // 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; + } + + // 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 ( 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( RtAudioError::WARNING ); + // Continue on to input section. + } + } + + if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) { + + // 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; + } + + // Read samples from device. + result = read( handle->id[1], buffer, samples * formatBytes(format) ); + + 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( RtAudioError::WARNING ); + goto unlock; + } + + // Do byte swapping if necessary. + if ( stream_.doByteSwap[1] ) + byteSwapBuffer( buffer, samples, format ); + + // Do buffer conversion if necessary. + if ( stream_.doConvertBuffer[1] ) + convertBuffer( stream_.userBuffer[1], stream_.deviceBuffer, stream_.convertInfo[1] ); + } + + unlock: + MUTEX_UNLOCK( &stream_.mutex ); + + RtApi::tickStreamTime(); + if ( doStopStream == 1 ) this->stopStream(); +} + +static void *ossCallbackHandler( void *ptr ) +{ + CallbackInfo *info = (CallbackInfo *) ptr; + RtApiOss *object = (RtApiOss *) info->object; + bool *isRunning = &info->isRunning; + + while ( *isRunning == true ) { + pthread_testcancel(); + object->callbackEvent(); + } + + pthread_exit( NULL ); +} + +//******************** End of __LINUX_OSS__ *********************// +#endif + + +// *************************************************** // +// +// Protected common (OS-independent) RtAudio methods. +// +// *************************************************** // + +// This method can be modified to control the behavior of error +// message printing. +void RtApi :: error( RtAudioError::Type type ) +{ + errorStream_.str(""); // clear the ostringstream + + RtAudioErrorCallback errorCallback = (RtAudioErrorCallback) stream_.callbackInfo.errorCallback; + if ( errorCallback ) { + // abortStream() can generate new error messages. Ignore them. Just keep original one. + + if ( firstErrorOccurred_ ) + return; + + firstErrorOccurred_ = true; + const std::string errorMessage = errorText_; + + if ( type != RtAudioError::WARNING && stream_.state != STREAM_STOPPED) { + stream_.callbackInfo.isRunning = false; // exit from the thread + abortStream(); + } + + errorCallback( type, errorMessage ); + firstErrorOccurred_ = false; + return; + } + + if ( type == RtAudioError::WARNING && showWarnings_ == true ) + std::cerr << '\n' << errorText_ << "\n\n"; + else if ( type != RtAudioError::WARNING ) + throw( RtAudioError( errorText_, type ) ); +} + +void RtApi :: verifyStream() +{ + if ( stream_.state == STREAM_CLOSED ) { + errorText_ = "RtApi:: a stream is not open!"; + error( RtAudioError::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; + stream_.callbackInfo.errorCallback = 0; + 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_SINT32 || format == RTAUDIO_FLOAT32 ) + return 4; + else if ( format == RTAUDIO_FLOAT64 ) + return 8; + else if ( format == RTAUDIO_SINT24 ) + return 3; + else if ( format == RTAUDIO_SINT8 ) + return 1; + + errorText_ = "RtApi::formatBytes: undefined format."; + error( RtAudioError::WARNING ); + + 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]; + } + + 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 { + 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 { // 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_.bufferSize ); + stream_.convertInfo[mode].outOffset.push_back( k * stream_.bufferSize ); + stream_.convertInfo[mode].inJump = 1; + stream_.convertInfo[mode].outJump = 1; + } + } + } + + // 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 { + 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 lower 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; + } + } + 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; + } + } + else if (info.inFormat == RTAUDIO_SINT24) { + Int24 *in = (Int24 *)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]].asInt()); + out[info.outOffset[j]] += 0.5; + out[info.outOffset[j]] *= scale; + } + 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_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.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; + } + } + else if (info.inFormat == RTAUDIO_SINT24) { + Int24 *in = (Int24 *)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]].asInt()); + out[info.outOffset[j]] += 0.5; + out[info.outOffset[j]] *= scale; + } + 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_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.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) { + Int24 *in = (Int24 *)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]].asInt(); + 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_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.outFormat == RTAUDIO_SINT24) { + Int24 *out = (Int24 *)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]] << 16); + //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]] << 8); + //out[info.outOffset[j]] <<= 8; + } + in += info.inJump; + out += info.outJump; + } + } + else if (info.inFormat == RTAUDIO_SINT24) { + // Channel compensation and/or (de)interleaving only. + Int24 *in = (Int24 *)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]] >> 8); + //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_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.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) { + Int24 *in = (Int24 *)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]].asInt() >> 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]] = (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_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; + } + } + 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) { + Int24 *in = (Int24 *)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]].asInt() >> 16); + } + 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_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_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_SINT24 ) { + for ( unsigned int i=0; i<samples; i++ ) { + // Swap 1st and 3rd bytes. + val = *(ptr); + *(ptr) = *(ptr+2); + *(ptr+2) = val; + + // Increment 2 more bytes. + ptr += 2; + } + } + 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 + // + // Local Variables: + // c-basic-offset: 2 + // indent-tabs-mode: nil + // End: + // + // vim: et sts=2 sw=2 + +#endif |