|
- /************************************************************************/
- /*! \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) 2001-2017 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 5.0.0
-
- #include "RtAudio.h"
- #include <iostream>
- #include <cstdlib>
- #include <cstring>
- #include <climits>
- #include <cmath>
- #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__)
- #define MUTEX_INITIALIZE(A) InitializeCriticalSection(A)
- #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 )
- {
- return RTAUDIO_VERSION;
- }
-
- // Define API names and display names.
- // Must be in same order as API enum.
- extern "C" {
- const char* rtaudio_api_names[][2] = {
- { "unspecified" , "Unknown" },
- { "alsa" , "ALSA" },
- { "pulse" , "Pulse" },
- { "oss" , "OpenSoundSystem" },
- { "jack" , "Jack" },
- { "core" , "CoreAudio" },
- { "wasapi" , "WASAPI" },
- { "asio" , "ASIO" },
- { "ds" , "DirectSound" },
- { "dummy" , "Dummy" },
- };
- const unsigned int rtaudio_num_api_names =
- sizeof(rtaudio_api_names)/sizeof(rtaudio_api_names[0]);
-
- // The order here will control the order of RtAudio's API search in
- // the constructor.
- extern "C" const RtAudio::Api rtaudio_compiled_apis[] = {
- #if defined(__UNIX_JACK__)
- RtAudio::UNIX_JACK,
- #endif
- #if defined(__LINUX_PULSE__)
- RtAudio::LINUX_PULSE,
- #endif
- #if defined(__LINUX_ALSA__)
- RtAudio::LINUX_ALSA,
- #endif
- #if defined(__LINUX_OSS__)
- RtAudio::LINUX_OSS,
- #endif
- #if defined(__WINDOWS_ASIO__)
- RtAudio::WINDOWS_ASIO,
- #endif
- #if defined(__WINDOWS_WASAPI__)
- RtAudio::WINDOWS_WASAPI,
- #endif
- #if defined(__WINDOWS_DS__)
- RtAudio::WINDOWS_DS,
- #endif
- #if defined(__MACOSX_CORE__)
- RtAudio::MACOSX_CORE,
- #endif
- #if defined(__RTAUDIO_DUMMY__)
- RtAudio::RTAUDIO_DUMMY,
- #endif
- RtAudio::UNSPECIFIED,
- };
- extern "C" const unsigned int rtaudio_num_compiled_apis =
- sizeof(rtaudio_compiled_apis)/sizeof(rtaudio_compiled_apis[0])-1;
- }
-
- // This is a compile-time check that rtaudio_num_api_names == RtAudio::NUM_APIS.
- // If the build breaks here, check that they match.
- template<bool b> class StaticAssert { private: StaticAssert() {} };
- template<> class StaticAssert<true>{ public: StaticAssert() {} };
- class StaticAssertions { StaticAssertions() {
- StaticAssert<rtaudio_num_api_names == RtAudio::NUM_APIS>();
- }};
-
- void RtAudio :: getCompiledApi( std::vector<RtAudio::Api> &apis )
- {
- apis = std::vector<RtAudio::Api>(rtaudio_compiled_apis,
- rtaudio_compiled_apis + rtaudio_num_compiled_apis);
- }
-
- std::string RtAudio :: getApiName( RtAudio::Api api )
- {
- if (api < 0 || api >= RtAudio::NUM_APIS)
- return "";
- return rtaudio_api_names[api][0];
- }
-
- std::string RtAudio :: getApiDisplayName( RtAudio::Api api )
- {
- if (api < 0 || api >= RtAudio::NUM_APIS)
- return "Unknown";
- return rtaudio_api_names[api][1];
- }
-
- RtAudio::Api RtAudio :: getCompiledApiByName( const std::string &name )
- {
- unsigned int i=0;
- for (i = 0; i < rtaudio_num_compiled_apis; ++i)
- if (name == rtaudio_api_names[rtaudio_compiled_apis[i]][0])
- return rtaudio_compiled_apis[i];
- return RtAudio::UNSPECIFIED;
- }
-
- 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()
- {
- 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;
- #if defined( HAVE_GETTIMEOFDAY )
- gettimeofday( &stream_.lastTickTimestamp, NULL );
- #endif
- }
-
- 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; }
- };
-
- #if !defined(__RTAUDIO_DEBUG__)
- static void jackSilentError( const char * ) {};
- #endif
-
- RtApiJack :: RtApiJack()
- :shouldAutoconnect_(true) {
- // 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, JACK_DEFAULT_AUDIO_TYPE, 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, JACK_DEFAULT_AUDIO_TYPE, 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(), JACK_DEFAULT_AUDIO_TYPE, 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(), JACK_DEFAULT_AUDIO_TYPE, 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, JACK_DEFAULT_AUDIO_TYPE, 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;
- }
-
- unsigned long flag = JackPortIsInput;
- if ( mode == INPUT ) flag = JackPortIsOutput;
-
- if ( ! (options && (options->flags & RTAUDIO_JACK_DONT_CONNECT)) ) {
- // 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, deviceName.c_str(), JACK_DEFAULT_AUDIO_TYPE, 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(), JACK_DEFAULT_AUDIO_TYPE, 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 *) &stream_.apiHandle );
- 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 );
-
- if ( options && options->flags & RTAUDIO_JACK_DONT_CONNECT ) shouldAutoconnect_ = false;
-
- 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 ( shouldAutoconnect_ && (stream_.mode == OUTPUT || stream_.mode == DUPLEX) ) {
- result = 1;
- ports = jack_get_ports( handle->client, handle->deviceName[0].c_str(), JACK_DEFAULT_AUDIO_TYPE, 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 ( shouldAutoconnect_ && (stream_.mode == INPUT || stream_.mode == DUPLEX) ) {
- result = 1;
- ports = jack_get_ports( handle->client, handle->deviceName[1].c_str(), JACK_DEFAULT_AUDIO_TYPE, 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 (e.g. Creative's 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 <mfapi.h>
- #include <mferror.h>
- #include <mfplay.h>
- #include <mftransform.h>
- #include <wmcodecdsp.h>
-
- #include <audioclient.h>
- #include <avrt.h>
- #include <mmdeviceapi.h>
- #include <functiondiscoverykeys_devpkey.h>
-
- #ifndef MF_E_TRANSFORM_NEED_MORE_INPUT
- #define MF_E_TRANSFORM_NEED_MORE_INPUT _HRESULT_TYPEDEF_(0xc00d6d72)
- #endif
-
- #ifndef MFSTARTUP_NOSOCKET
- #define MFSTARTUP_NOSOCKET 0x1
- #endif
-
- #ifdef _MSC_VER
- #pragma comment( lib, "ksuser" )
- #pragma comment( lib, "mfplat.lib" )
- #pragma comment( lib, "mfuuid.lib" )
- #pragma comment( lib, "wmcodecdspuuid" )
- #endif
-
- //=============================================================================
-
- #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 WasapiResampler class is used to perform this conversion between
- // HwIn->UserIn and UserOut->HwOut during the stream callback loop.
- class WasapiResampler
- {
- public:
- WasapiResampler( bool isFloat, unsigned int bitsPerSample, unsigned int channelCount,
- unsigned int inSampleRate, unsigned int outSampleRate )
- : _bytesPerSample( bitsPerSample / 8 )
- , _channelCount( channelCount )
- , _sampleRatio( ( float ) outSampleRate / inSampleRate )
- , _transformUnk( NULL )
- , _transform( NULL )
- , _mediaType( NULL )
- , _inputMediaType( NULL )
- , _outputMediaType( NULL )
-
- #ifdef __IWMResamplerProps_FWD_DEFINED__
- , _resamplerProps( NULL )
- #endif
- {
- // 1. Initialization
-
- MFStartup( MF_VERSION, MFSTARTUP_NOSOCKET );
-
- // 2. Create Resampler Transform Object
-
- CoCreateInstance( CLSID_CResamplerMediaObject, NULL, CLSCTX_INPROC_SERVER,
- IID_IUnknown, ( void** ) &_transformUnk );
-
- _transformUnk->QueryInterface( IID_PPV_ARGS( &_transform ) );
-
- #ifdef __IWMResamplerProps_FWD_DEFINED__
- _transformUnk->QueryInterface( IID_PPV_ARGS( &_resamplerProps ) );
- _resamplerProps->SetHalfFilterLength( 60 ); // best conversion quality
- #endif
-
- // 3. Specify input / output format
-
- MFCreateMediaType( &_mediaType );
- _mediaType->SetGUID( MF_MT_MAJOR_TYPE, MFMediaType_Audio );
- _mediaType->SetGUID( MF_MT_SUBTYPE, isFloat ? MFAudioFormat_Float : MFAudioFormat_PCM );
- _mediaType->SetUINT32( MF_MT_AUDIO_NUM_CHANNELS, channelCount );
- _mediaType->SetUINT32( MF_MT_AUDIO_SAMPLES_PER_SECOND, inSampleRate );
- _mediaType->SetUINT32( MF_MT_AUDIO_BLOCK_ALIGNMENT, _bytesPerSample * channelCount );
- _mediaType->SetUINT32( MF_MT_AUDIO_AVG_BYTES_PER_SECOND, _bytesPerSample * channelCount * inSampleRate );
- _mediaType->SetUINT32( MF_MT_AUDIO_BITS_PER_SAMPLE, bitsPerSample );
- _mediaType->SetUINT32( MF_MT_ALL_SAMPLES_INDEPENDENT, TRUE );
-
- MFCreateMediaType( &_inputMediaType );
- _mediaType->CopyAllItems( _inputMediaType );
-
- _transform->SetInputType( 0, _inputMediaType, 0 );
-
- MFCreateMediaType( &_outputMediaType );
- _mediaType->CopyAllItems( _outputMediaType );
-
- _outputMediaType->SetUINT32( MF_MT_AUDIO_SAMPLES_PER_SECOND, outSampleRate );
- _outputMediaType->SetUINT32( MF_MT_AUDIO_AVG_BYTES_PER_SECOND, _bytesPerSample * channelCount * outSampleRate );
-
- _transform->SetOutputType( 0, _outputMediaType, 0 );
-
- // 4. Send stream start messages to Resampler
-
- _transform->ProcessMessage( MFT_MESSAGE_COMMAND_FLUSH, 0 );
- _transform->ProcessMessage( MFT_MESSAGE_NOTIFY_BEGIN_STREAMING, 0 );
- _transform->ProcessMessage( MFT_MESSAGE_NOTIFY_START_OF_STREAM, 0 );
- }
-
- ~WasapiResampler()
- {
- // 8. Send stream stop messages to Resampler
-
- _transform->ProcessMessage( MFT_MESSAGE_NOTIFY_END_OF_STREAM, 0 );
- _transform->ProcessMessage( MFT_MESSAGE_NOTIFY_END_STREAMING, 0 );
-
- // 9. Cleanup
-
- MFShutdown();
-
- SAFE_RELEASE( _transformUnk );
- SAFE_RELEASE( _transform );
- SAFE_RELEASE( _mediaType );
- SAFE_RELEASE( _inputMediaType );
- SAFE_RELEASE( _outputMediaType );
-
- #ifdef __IWMResamplerProps_FWD_DEFINED__
- SAFE_RELEASE( _resamplerProps );
- #endif
- }
-
- void Convert( char* outBuffer, const char* inBuffer, unsigned int inSampleCount, unsigned int& outSampleCount )
- {
- unsigned int inputBufferSize = _bytesPerSample * _channelCount * inSampleCount;
- if ( _sampleRatio == 1 )
- {
- // no sample rate conversion required
- memcpy( outBuffer, inBuffer, inputBufferSize );
- outSampleCount = inSampleCount;
- return;
- }
-
- unsigned int outputBufferSize = ( unsigned int ) ceilf( inputBufferSize * _sampleRatio ) + ( _bytesPerSample * _channelCount );
-
- IMFMediaBuffer* rInBuffer;
- IMFSample* rInSample;
- BYTE* rInByteBuffer = NULL;
-
- // 5. Create Sample object from input data
-
- MFCreateMemoryBuffer( inputBufferSize, &rInBuffer );
-
- rInBuffer->Lock( &rInByteBuffer, NULL, NULL );
- memcpy( rInByteBuffer, inBuffer, inputBufferSize );
- rInBuffer->Unlock();
- rInByteBuffer = NULL;
-
- rInBuffer->SetCurrentLength( inputBufferSize );
-
- MFCreateSample( &rInSample );
- rInSample->AddBuffer( rInBuffer );
-
- // 6. Pass input data to Resampler
-
- _transform->ProcessInput( 0, rInSample, 0 );
-
- SAFE_RELEASE( rInBuffer );
- SAFE_RELEASE( rInSample );
-
- // 7. Perform sample rate conversion
-
- IMFMediaBuffer* rOutBuffer = NULL;
- BYTE* rOutByteBuffer = NULL;
-
- MFT_OUTPUT_DATA_BUFFER rOutDataBuffer;
- DWORD rStatus;
- DWORD rBytes = outputBufferSize; // maximum bytes accepted per ProcessOutput
-
- // 7.1 Create Sample object for output data
-
- memset( &rOutDataBuffer, 0, sizeof rOutDataBuffer );
- MFCreateSample( &( rOutDataBuffer.pSample ) );
- MFCreateMemoryBuffer( rBytes, &rOutBuffer );
- rOutDataBuffer.pSample->AddBuffer( rOutBuffer );
- rOutDataBuffer.dwStreamID = 0;
- rOutDataBuffer.dwStatus = 0;
- rOutDataBuffer.pEvents = NULL;
-
- // 7.2 Get output data from Resampler
-
- if ( _transform->ProcessOutput( 0, 1, &rOutDataBuffer, &rStatus ) == MF_E_TRANSFORM_NEED_MORE_INPUT )
- {
- outSampleCount = 0;
- SAFE_RELEASE( rOutBuffer );
- SAFE_RELEASE( rOutDataBuffer.pSample );
- return;
- }
-
- // 7.3 Write output data to outBuffer
-
- SAFE_RELEASE( rOutBuffer );
- rOutDataBuffer.pSample->ConvertToContiguousBuffer( &rOutBuffer );
- rOutBuffer->GetCurrentLength( &rBytes );
-
- rOutBuffer->Lock( &rOutByteBuffer, NULL, NULL );
- memcpy( outBuffer, rOutByteBuffer, rBytes );
- rOutBuffer->Unlock();
- rOutByteBuffer = NULL;
-
- outSampleCount = rBytes / _bytesPerSample / _channelCount;
- SAFE_RELEASE( rOutBuffer );
- SAFE_RELEASE( rOutDataBuffer.pSample );
- }
-
- private:
- unsigned int _bytesPerSample;
- unsigned int _channelCount;
- float _sampleRatio;
-
- IUnknown* _transformUnk;
- IMFTransform* _transform;
- IMFMediaType* _mediaType;
- IMFMediaType* _inputMediaType;
- IMFMediaType* _outputMediaType;
-
- #ifdef __IWMResamplerProps_FWD_DEFINED__
- IWMResamplerProps* _resamplerProps;
- #endif
- };
-
- //-----------------------------------------------------------------------------
-
- // 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 this runs on an old Windows, it will fail. Ignore and proceed.
- if ( FAILED( hr ) )
- deviceEnumerator_ = NULL;
- }
-
- //-----------------------------------------------------------------------------
-
- 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;
-
- if ( !deviceEnumerator_ )
- return 0;
-
- // 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;
- }
-
- // if device index falls within capture 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 capture device audio client.";
- goto Exit;
- }
-
- hr = captureAudioClient->GetMixFormat( &deviceFormat );
- if ( FAILED( hr ) ) {
- errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve capture device mix format.";
- goto Exit;
- }
-
- stream_.nDeviceChannels[mode] = deviceFormat->nChannels;
- captureAudioClient->GetStreamLatency( ( long long* ) &stream_.latency[mode] );
- }
-
- // if device index falls within render devices and is configured for loopback
- if ( device < renderDeviceCount && mode == INPUT )
- {
- // retrieve captureAudioClient from devicePtr
- IAudioClient*& captureAudioClient = ( ( WasapiHandle* ) stream_.apiHandle )->captureAudioClient;
-
- 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** ) &captureAudioClient );
- if ( FAILED( hr ) ) {
- errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve render device audio client.";
- goto Exit;
- }
-
- hr = captureAudioClient->GetMixFormat( &deviceFormat );
- if ( FAILED( hr ) ) {
- errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve render device mix format.";
- goto Exit;
- }
-
- stream_.nDeviceChannels[mode] = deviceFormat->nChannels;
- captureAudioClient->GetStreamLatency( ( long long* ) &stream_.latency[mode] );
- }
-
- // if device index falls within render devices and is configured for output
- if ( device < renderDeviceCount && mode == OUTPUT )
- {
- // 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 render device audio client.";
- goto Exit;
- }
-
- hr = renderAudioClient->GetMixFormat( &deviceFormat );
- if ( FAILED( hr ) ) {
- errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve render 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[0] != stream_.nDeviceChannels[0] ||
- stream_.nUserChannels[1] != stream_.nDeviceChannels[1] )
- 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;
- WasapiResampler* captureResampler = NULL;
- WasapiResampler* renderResampler = NULL;
-
- // 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 loopbackEnabled = stream_.device[INPUT] == stream_.device[OUTPUT];
- bool callbackPushed = true;
- 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;
- }
-
- // init captureResampler
- captureResampler = new WasapiResampler( stream_.deviceFormat[INPUT] == RTAUDIO_FLOAT32 || stream_.deviceFormat[INPUT] == RTAUDIO_FLOAT64,
- formatBytes( stream_.deviceFormat[INPUT] ) * 8, stream_.nDeviceChannels[INPUT],
- captureFormat->nSamplesPerSec, stream_.sampleRate );
-
- captureSrRatio = ( ( float ) captureFormat->nSamplesPerSec / stream_.sampleRate );
-
- if ( !captureClient ) {
- hr = captureAudioClient->Initialize( AUDCLNT_SHAREMODE_SHARED,
- loopbackEnabled ? AUDCLNT_STREAMFLAGS_LOOPBACK : AUDCLNT_STREAMFLAGS_EVENTCALLBACK,
- 0,
- 0,
- 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;
- }
-
- // don't configure captureEvent if in loopback mode
- if ( !loopbackEnabled )
- {
- // 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 )->captureEvent = captureEvent;
- }
-
- ( ( WasapiHandle* ) stream_.apiHandle )->captureClient = captureClient;
- }
-
- 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 ) ceilf( 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;
- }
-
- // init renderResampler
- renderResampler = new WasapiResampler( stream_.deviceFormat[OUTPUT] == RTAUDIO_FLOAT32 || stream_.deviceFormat[OUTPUT] == RTAUDIO_FLOAT64,
- formatBytes( stream_.deviceFormat[OUTPUT] ) * 8, stream_.nDeviceChannels[OUTPUT],
- stream_.sampleRate, renderFormat->nSamplesPerSec );
-
- renderSrRatio = ( ( float ) renderFormat->nSamplesPerSec / stream_.sampleRate );
-
- if ( !renderClient ) {
- hr = renderAudioClient->Initialize( AUDCLNT_SHAREMODE_SHARED,
- AUDCLNT_STREAMFLAGS_EVENTCALLBACK,
- 0,
- 0,
- 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 ) ceilf( 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;
- }
- }
-
- // malloc buffer memory
- if ( stream_.mode == INPUT )
- {
- using namespace std; // for ceilf
- convBuffSize = ( size_t ) ( ceilf( 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 ) ( ceilf( 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 ) ( ceilf( stream_.bufferSize * captureSrRatio ) ) * stream_.nDeviceChannels[INPUT] * formatBytes( stream_.deviceFormat[INPUT] ),
- ( size_t ) ( ceilf( 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] ) );
- }
-
- convBuffSize *= 2; // allow overflow for *SrRatio remainders
- 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 )
- {
- int samplesToPull = ( unsigned int ) floorf( stream_.bufferSize * captureSrRatio );
- if ( captureSrRatio != 1 )
- {
- // account for remainders
- samplesToPull--;
- }
-
- convBufferSize = 0;
- while ( convBufferSize < stream_.bufferSize )
- {
- // Pull callback buffer from inputBuffer
- callbackPulled = captureBuffer.pullBuffer( convBuffer,
- samplesToPull * stream_.nDeviceChannels[INPUT],
- stream_.deviceFormat[INPUT] );
-
- if ( !callbackPulled )
- {
- break;
- }
-
- // Convert callback buffer to user sample rate
- unsigned int deviceBufferOffset = convBufferSize * stream_.nDeviceChannels[INPUT] * formatBytes( stream_.deviceFormat[INPUT] );
- unsigned int convSamples = 0;
-
- captureResampler->Convert( stream_.deviceBuffer + deviceBufferOffset,
- convBuffer,
- samplesToPull,
- convSamples );
-
- convBufferSize += convSamples;
- samplesToPull = 1; // now pull one sample at a time until we have stream_.bufferSize samples
- }
-
- if ( callbackPulled )
- {
- 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 the last call to renderBuffer.PushBuffer() was successful
- if ( callbackPushed || convBufferSize == 0 )
- {
- 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
- renderResampler->Convert( convBuffer,
- stream_.deviceBuffer,
- stream_.bufferSize,
- convBufferSize );
- }
-
- // 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( loopbackEnabled ? renderEvent : 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 ) {
- // unsetting the callbackPulled flag lets the stream know that
- // the audio device is ready for another callback output buffer.
- callbackPulled = false;
-
- // tick stream time
- RtApi::tickStreamTime();
- }
-
- }
-
- Exit:
- // clean up
- CoTaskMemFree( captureFormat );
- CoTaskMemFree( renderFormat );
-
- free ( convBuffer );
- delete renderResampler;
- delete captureResampler;
-
- CoUninitialize();
-
- if ( !errorText_.empty() )
- error( errorType );
-
- // update stream state
- stream_.state = STREAM_STOPPED;
- }
-
- //******************** 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 <windows.h>
- #include <process.h>
- #include <mmsystem.h>
- #include <mmreg.h>
- #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 ( stream_.state != STREAM_CLOSED ) closeStream();
- if ( coInitialized_ ) CoUninitialize(); // balanced call.
- }
-
- // 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();
- MUTEX_UNLOCK( &stream_.mutex );
- 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 );
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