+++ /dev/null
-From: =?utf-8?q?IOhannes_m_zm=C3=B6lnig?= <umlaeute@debian.org>
-Date: Wed, 25 Oct 2017 14:21:33 +0200
-Subject: updated bundled and hacked RtAudio to RtAudio5
-
----
- src/core/kernelAudio.cpp | 2 +-
- src/deps/rtaudio-mod/RtAudio.cpp | 20580 +++++++++++++++++++------------------
- src/deps/rtaudio-mod/RtAudio.h | 113 +-
- 3 files changed, 10401 insertions(+), 10294 deletions(-)
-
---- giada.orig/src/core/kernelAudio.cpp
-+++ giada/src/core/kernelAudio.cpp
-@@ -55,7 +55,7 @@
-
- jack_client_t* jackGetHandle()
- {
-- return static_cast<jack_client_t*>(rtSystem->rtapi_->__HACK__getJackClient());
-+ return static_cast<jack_client_t*>(rtSystem->GIADA_HACK__getJackClient());
- }
-
- #endif
---- giada.orig/src/deps/rtaudio-mod/RtAudio.cpp
-+++ giada/src/deps/rtaudio-mod/RtAudio.cpp
-@@ -1,10237 +1,10343 @@
--/************************************************************************/\r
--/*! \class RtAudio\r
-- \brief Realtime audio i/o C++ classes.\r
--\r
-- RtAudio provides a common API (Application Programming Interface)\r
-- for realtime audio input/output across Linux (native ALSA, Jack,\r
-- and OSS), Macintosh OS X (CoreAudio and Jack), and Windows\r
-- (DirectSound, ASIO and WASAPI) operating systems.\r
--\r
-- RtAudio WWW site: http://www.music.mcgill.ca/~gary/rtaudio/\r
--\r
-- RtAudio: realtime audio i/o C++ classes\r
-- Copyright (c) 2001-2016 Gary P. Scavone\r
--\r
-- Permission is hereby granted, free of charge, to any person\r
-- obtaining a copy of this software and associated documentation files\r
-- (the "Software"), to deal in the Software without restriction,\r
-- including without limitation the rights to use, copy, modify, merge,\r
-- publish, distribute, sublicense, and/or sell copies of the Software,\r
-- and to permit persons to whom the Software is furnished to do so,\r
-- subject to the following conditions:\r
--\r
-- The above copyright notice and this permission notice shall be\r
-- included in all copies or substantial portions of the Software.\r
--\r
-- Any person wishing to distribute modifications to the Software is\r
-- asked to send the modifications to the original developer so that\r
-- they can be incorporated into the canonical version. This is,\r
-- however, not a binding provision of this license.\r
--\r
-- THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND,\r
-- EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF\r
-- MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT.\r
-- IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR\r
-- ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF\r
-- CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION\r
-- WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.\r
--*/\r
--/************************************************************************/\r
--\r
--// RtAudio: Version 4.1.2\r
--\r
--#include "RtAudio.h"\r
--#include <iostream>\r
--#include <cstdlib>\r
--#include <cstring>\r
--#include <climits>\r
--#include <algorithm>\r
--\r
--// Static variable definitions.\r
--const unsigned int RtApi::MAX_SAMPLE_RATES = 14;\r
--const unsigned int RtApi::SAMPLE_RATES[] = {\r
-- 4000, 5512, 8000, 9600, 11025, 16000, 22050,\r
-- 32000, 44100, 48000, 88200, 96000, 176400, 192000\r
--};\r
--\r
--#if defined(__WINDOWS_DS__) || defined(__WINDOWS_ASIO__) || defined(__WINDOWS_WASAPI__)\r
-- #define MUTEX_INITIALIZE(A) InitializeCriticalSection(A)\r
-- #define MUTEX_DESTROY(A) DeleteCriticalSection(A)\r
-- #define MUTEX_LOCK(A) EnterCriticalSection(A)\r
-- #define MUTEX_UNLOCK(A) LeaveCriticalSection(A)\r
--\r
-- #include "tchar.h"\r
--\r
-- static std::string convertCharPointerToStdString(const char *text)\r
-- {\r
-- return std::string(text);\r
-- }\r
--\r
-- static std::string convertCharPointerToStdString(const wchar_t *text)\r
-- {\r
-- int length = WideCharToMultiByte(CP_UTF8, 0, text, -1, NULL, 0, NULL, NULL);\r
-- std::string s( length-1, '\0' );\r
-- WideCharToMultiByte(CP_UTF8, 0, text, -1, &s[0], length, NULL, NULL);\r
-- return s;\r
-- }\r
--\r
--#elif defined(__LINUX_ALSA__) || defined(__LINUX_PULSE__) || defined(__UNIX_JACK__) || defined(__LINUX_OSS__) || defined(__MACOSX_CORE__)\r
-- // pthread API\r
-- #define MUTEX_INITIALIZE(A) pthread_mutex_init(A, NULL)\r
-- #define MUTEX_DESTROY(A) pthread_mutex_destroy(A)\r
-- #define MUTEX_LOCK(A) pthread_mutex_lock(A)\r
-- #define MUTEX_UNLOCK(A) pthread_mutex_unlock(A)\r
--#else\r
-- #define MUTEX_INITIALIZE(A) abs(*A) // dummy definitions\r
-- #define MUTEX_DESTROY(A) abs(*A) // dummy definitions\r
--#endif\r
--\r
--// *************************************************** //\r
--//\r
--// RtAudio definitions.\r
--//\r
--// *************************************************** //\r
--\r
--std::string RtAudio :: getVersion( void ) throw()\r
--{\r
-- return RTAUDIO_VERSION;\r
--}\r
--\r
--void RtAudio :: getCompiledApi( std::vector<RtAudio::Api> &apis ) throw()\r
--{\r
-- apis.clear();\r
--\r
-- // The order here will control the order of RtAudio's API search in\r
-- // the constructor.\r
--#if defined(__UNIX_JACK__)\r
-- apis.push_back( UNIX_JACK );\r
--#endif\r
--#if defined(__LINUX_ALSA__)\r
-- apis.push_back( LINUX_ALSA );\r
--#endif\r
--#if defined(__LINUX_PULSE__)\r
-- apis.push_back( LINUX_PULSE );\r
--#endif\r
--#if defined(__LINUX_OSS__)\r
-- apis.push_back( LINUX_OSS );\r
--#endif\r
--#if defined(__WINDOWS_ASIO__)\r
-- apis.push_back( WINDOWS_ASIO );\r
--#endif\r
--#if defined(__WINDOWS_WASAPI__)\r
-- apis.push_back( WINDOWS_WASAPI );\r
--#endif\r
--#if defined(__WINDOWS_DS__)\r
-- apis.push_back( WINDOWS_DS );\r
--#endif\r
--#if defined(__MACOSX_CORE__)\r
-- apis.push_back( MACOSX_CORE );\r
--#endif\r
--#if defined(__RTAUDIO_DUMMY__)\r
-- apis.push_back( RTAUDIO_DUMMY );\r
--#endif\r
--}\r
--\r
--void RtAudio :: openRtApi( RtAudio::Api api )\r
--{\r
-- if ( rtapi_ )\r
-- delete rtapi_;\r
-- rtapi_ = 0;\r
--\r
--#if defined(__UNIX_JACK__)\r
-- if ( api == UNIX_JACK )\r
-- rtapi_ = new RtApiJack();\r
--#endif\r
--#if defined(__LINUX_ALSA__)\r
-- if ( api == LINUX_ALSA )\r
-- rtapi_ = new RtApiAlsa();\r
--#endif\r
--#if defined(__LINUX_PULSE__)\r
-- if ( api == LINUX_PULSE )\r
-- rtapi_ = new RtApiPulse();\r
--#endif\r
--#if defined(__LINUX_OSS__)\r
-- if ( api == LINUX_OSS )\r
-- rtapi_ = new RtApiOss();\r
--#endif\r
--#if defined(__WINDOWS_ASIO__)\r
-- if ( api == WINDOWS_ASIO )\r
-- rtapi_ = new RtApiAsio();\r
--#endif\r
--#if defined(__WINDOWS_WASAPI__)\r
-- if ( api == WINDOWS_WASAPI )\r
-- rtapi_ = new RtApiWasapi();\r
--#endif\r
--#if defined(__WINDOWS_DS__)\r
-- if ( api == WINDOWS_DS )\r
-- rtapi_ = new RtApiDs();\r
--#endif\r
--#if defined(__MACOSX_CORE__)\r
-- if ( api == MACOSX_CORE )\r
-- rtapi_ = new RtApiCore();\r
--#endif\r
--#if defined(__RTAUDIO_DUMMY__)\r
-- if ( api == RTAUDIO_DUMMY )\r
-- rtapi_ = new RtApiDummy();\r
--#endif\r
--}\r
--\r
--RtAudio :: RtAudio( RtAudio::Api api )\r
--{\r
-- rtapi_ = 0;\r
--\r
-- if ( api != UNSPECIFIED ) {\r
-- // Attempt to open the specified API.\r
-- openRtApi( api );\r
-- if ( rtapi_ ) return;\r
--\r
-- // No compiled support for specified API value. Issue a debug\r
-- // warning and continue as if no API was specified.\r
-- std::cerr << "\nRtAudio: no compiled support for specified API argument!\n" << std::endl;\r
-- }\r
--\r
-- // Iterate through the compiled APIs and return as soon as we find\r
-- // one with at least one device or we reach the end of the list.\r
-- std::vector< RtAudio::Api > apis;\r
-- getCompiledApi( apis );\r
-- for ( unsigned int i=0; i<apis.size(); i++ ) {\r
-- openRtApi( apis[i] );\r
-- if ( rtapi_ && rtapi_->getDeviceCount() ) break;\r
-- }\r
--\r
-- if ( rtapi_ ) return;\r
--\r
-- // It should not be possible to get here because the preprocessor\r
-- // definition __RTAUDIO_DUMMY__ is automatically defined if no\r
-- // API-specific definitions are passed to the compiler. But just in\r
-- // case something weird happens, we'll thow an error.\r
-- std::string errorText = "\nRtAudio: no compiled API support found ... critical error!!\n\n";\r
-- throw( RtAudioError( errorText, RtAudioError::UNSPECIFIED ) );\r
--}\r
--\r
--RtAudio :: ~RtAudio() throw()\r
--{\r
-- if ( rtapi_ )\r
-- delete rtapi_;\r
--}\r
--\r
--void RtAudio :: openStream( RtAudio::StreamParameters *outputParameters,\r
-- RtAudio::StreamParameters *inputParameters,\r
-- RtAudioFormat format, unsigned int sampleRate,\r
-- unsigned int *bufferFrames,\r
-- RtAudioCallback callback, void *userData,\r
-- RtAudio::StreamOptions *options,\r
-- RtAudioErrorCallback errorCallback )\r
--{\r
-- return rtapi_->openStream( outputParameters, inputParameters, format,\r
-- sampleRate, bufferFrames, callback,\r
-- userData, options, errorCallback );\r
--}\r
--\r
--// *************************************************** //\r
--//\r
--// Public RtApi definitions (see end of file for\r
--// private or protected utility functions).\r
--//\r
--// *************************************************** //\r
--\r
--RtApi :: RtApi()\r
--{\r
-- stream_.state = STREAM_CLOSED;\r
-- stream_.mode = UNINITIALIZED;\r
-- stream_.apiHandle = 0;\r
-- stream_.userBuffer[0] = 0;\r
-- stream_.userBuffer[1] = 0;\r
-- MUTEX_INITIALIZE( &stream_.mutex );\r
-- showWarnings_ = true;\r
-- firstErrorOccurred_ = false;\r
--}\r
--\r
--RtApi :: ~RtApi()\r
--{\r
-- MUTEX_DESTROY( &stream_.mutex );\r
--}\r
--\r
--void RtApi :: openStream( RtAudio::StreamParameters *oParams,\r
-- RtAudio::StreamParameters *iParams,\r
-- RtAudioFormat format, unsigned int sampleRate,\r
-- unsigned int *bufferFrames,\r
-- RtAudioCallback callback, void *userData,\r
-- RtAudio::StreamOptions *options,\r
-- RtAudioErrorCallback errorCallback )\r
--{\r
-- if ( stream_.state != STREAM_CLOSED ) {\r
-- errorText_ = "RtApi::openStream: a stream is already open!";\r
-- error( RtAudioError::INVALID_USE );\r
-- return;\r
-- }\r
--\r
-- // Clear stream information potentially left from a previously open stream.\r
-- clearStreamInfo();\r
--\r
-- if ( oParams && oParams->nChannels < 1 ) {\r
-- errorText_ = "RtApi::openStream: a non-NULL output StreamParameters structure cannot have an nChannels value less than one.";\r
-- error( RtAudioError::INVALID_USE );\r
-- return;\r
-- }\r
--\r
-- if ( iParams && iParams->nChannels < 1 ) {\r
-- errorText_ = "RtApi::openStream: a non-NULL input StreamParameters structure cannot have an nChannels value less than one.";\r
-- error( RtAudioError::INVALID_USE );\r
-- return;\r
-- }\r
--\r
-- if ( oParams == NULL && iParams == NULL ) {\r
-- errorText_ = "RtApi::openStream: input and output StreamParameters structures are both NULL!";\r
-- error( RtAudioError::INVALID_USE );\r
-- return;\r
-- }\r
--\r
-- if ( formatBytes(format) == 0 ) {\r
-- errorText_ = "RtApi::openStream: 'format' parameter value is undefined.";\r
-- error( RtAudioError::INVALID_USE );\r
-- return;\r
-- }\r
--\r
-- unsigned int nDevices = getDeviceCount();\r
-- unsigned int oChannels = 0;\r
-- if ( oParams ) {\r
-- oChannels = oParams->nChannels;\r
-- if ( oParams->deviceId >= nDevices ) {\r
-- errorText_ = "RtApi::openStream: output device parameter value is invalid.";\r
-- error( RtAudioError::INVALID_USE );\r
-- return;\r
-- }\r
-- }\r
--\r
-- unsigned int iChannels = 0;\r
-- if ( iParams ) {\r
-- iChannels = iParams->nChannels;\r
-- if ( iParams->deviceId >= nDevices ) {\r
-- errorText_ = "RtApi::openStream: input device parameter value is invalid.";\r
-- error( RtAudioError::INVALID_USE );\r
-- return;\r
-- }\r
-- }\r
--\r
-- bool result;\r
--\r
-- if ( oChannels > 0 ) {\r
--\r
-- result = probeDeviceOpen( oParams->deviceId, OUTPUT, oChannels, oParams->firstChannel,\r
-- sampleRate, format, bufferFrames, options );\r
-- if ( result == false ) {\r
-- error( RtAudioError::SYSTEM_ERROR );\r
-- return;\r
-- }\r
-- }\r
--\r
-- if ( iChannels > 0 ) {\r
--\r
-- result = probeDeviceOpen( iParams->deviceId, INPUT, iChannels, iParams->firstChannel,\r
-- sampleRate, format, bufferFrames, options );\r
-- if ( result == false ) {\r
-- if ( oChannels > 0 ) closeStream();\r
-- error( RtAudioError::SYSTEM_ERROR );\r
-- return;\r
-- }\r
-- }\r
--\r
-- stream_.callbackInfo.callback = (void *) callback;\r
-- stream_.callbackInfo.userData = userData;\r
-- stream_.callbackInfo.errorCallback = (void *) errorCallback;\r
--\r
-- if ( options ) options->numberOfBuffers = stream_.nBuffers;\r
-- stream_.state = STREAM_STOPPED;\r
--}\r
--\r
--unsigned int RtApi :: getDefaultInputDevice( void )\r
--{\r
-- // Should be implemented in subclasses if possible.\r
-- return 0;\r
--}\r
--\r
--unsigned int RtApi :: getDefaultOutputDevice( void )\r
--{\r
-- // Should be implemented in subclasses if possible.\r
-- return 0;\r
--}\r
--\r
--void RtApi :: closeStream( void )\r
--{\r
-- // MUST be implemented in subclasses!\r
-- return;\r
--}\r
--\r
--bool RtApi :: probeDeviceOpen( unsigned int /*device*/, StreamMode /*mode*/, unsigned int /*channels*/,\r
-- unsigned int /*firstChannel*/, unsigned int /*sampleRate*/,\r
-- RtAudioFormat /*format*/, unsigned int * /*bufferSize*/,\r
-- RtAudio::StreamOptions * /*options*/ )\r
--{\r
-- // MUST be implemented in subclasses!\r
-- return FAILURE;\r
--}\r
--\r
--void RtApi :: tickStreamTime( void )\r
--{\r
-- // Subclasses that do not provide their own implementation of\r
-- // getStreamTime should call this function once per buffer I/O to\r
-- // provide basic stream time support.\r
--\r
-- stream_.streamTime += ( stream_.bufferSize * 1.0 / stream_.sampleRate );\r
--\r
--#if defined( HAVE_GETTIMEOFDAY )\r
-- gettimeofday( &stream_.lastTickTimestamp, NULL );\r
--#endif\r
--}\r
--\r
--long RtApi :: getStreamLatency( void )\r
--{\r
-- verifyStream();\r
--\r
-- long totalLatency = 0;\r
-- if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX )\r
-- totalLatency = stream_.latency[0];\r
-- if ( stream_.mode == INPUT || stream_.mode == DUPLEX )\r
-- totalLatency += stream_.latency[1];\r
--\r
-- return totalLatency;\r
--}\r
--\r
--double RtApi :: getStreamTime( void )\r
--{\r
-- verifyStream();\r
--\r
--#if defined( HAVE_GETTIMEOFDAY )\r
-- // Return a very accurate estimate of the stream time by\r
-- // adding in the elapsed time since the last tick.\r
-- struct timeval then;\r
-- struct timeval now;\r
--\r
-- if ( stream_.state != STREAM_RUNNING || stream_.streamTime == 0.0 )\r
-- return stream_.streamTime;\r
--\r
-- gettimeofday( &now, NULL );\r
-- then = stream_.lastTickTimestamp;\r
-- return stream_.streamTime +\r
-- ((now.tv_sec + 0.000001 * now.tv_usec) -\r
-- (then.tv_sec + 0.000001 * then.tv_usec));\r
--#else\r
-- return stream_.streamTime;\r
--#endif\r
--}\r
--\r
--void RtApi :: setStreamTime( double time )\r
--{\r
-- verifyStream();\r
--\r
-- if ( time >= 0.0 )\r
-- stream_.streamTime = time;\r
--}\r
--\r
--unsigned int RtApi :: getStreamSampleRate( void )\r
--{\r
-- verifyStream();\r
--\r
-- return stream_.sampleRate;\r
--}\r
--\r
--\r
--// *************************************************** //\r
--//\r
--// OS/API-specific methods.\r
--//\r
--// *************************************************** //\r
--\r
--#if defined(__MACOSX_CORE__)\r
--\r
--// The OS X CoreAudio API is designed to use a separate callback\r
--// procedure for each of its audio devices. A single RtAudio duplex\r
--// stream using two different devices is supported here, though it\r
--// cannot be guaranteed to always behave correctly because we cannot\r
--// synchronize these two callbacks.\r
--//\r
--// A property listener is installed for over/underrun information.\r
--// However, no functionality is currently provided to allow property\r
--// listeners to trigger user handlers because it is unclear what could\r
--// be done if a critical stream parameter (buffer size, sample rate,\r
--// device disconnect) notification arrived. The listeners entail\r
--// quite a bit of extra code and most likely, a user program wouldn't\r
--// be prepared for the result anyway. However, we do provide a flag\r
--// to the client callback function to inform of an over/underrun.\r
--\r
--// A structure to hold various information related to the CoreAudio API\r
--// implementation.\r
--struct CoreHandle {\r
-- AudioDeviceID id[2]; // device ids\r
--#if defined( MAC_OS_X_VERSION_10_5 ) && ( MAC_OS_X_VERSION_MIN_REQUIRED >= MAC_OS_X_VERSION_10_5 )\r
-- AudioDeviceIOProcID procId[2];\r
--#endif\r
-- UInt32 iStream[2]; // device stream index (or first if using multiple)\r
-- UInt32 nStreams[2]; // number of streams to use\r
-- bool xrun[2];\r
-- char *deviceBuffer;\r
-- pthread_cond_t condition;\r
-- int drainCounter; // Tracks callback counts when draining\r
-- bool internalDrain; // Indicates if stop is initiated from callback or not.\r
--\r
-- CoreHandle()\r
-- :deviceBuffer(0), drainCounter(0), internalDrain(false) { nStreams[0] = 1; nStreams[1] = 1; id[0] = 0; id[1] = 0; xrun[0] = false; xrun[1] = false; }\r
--};\r
--\r
--RtApiCore:: RtApiCore()\r
--{\r
--#if defined( AVAILABLE_MAC_OS_X_VERSION_10_6_AND_LATER )\r
-- // This is a largely undocumented but absolutely necessary\r
-- // requirement starting with OS-X 10.6. If not called, queries and\r
-- // updates to various audio device properties are not handled\r
-- // correctly.\r
-- CFRunLoopRef theRunLoop = NULL;\r
-- AudioObjectPropertyAddress property = { kAudioHardwarePropertyRunLoop,\r
-- kAudioObjectPropertyScopeGlobal,\r
-- kAudioObjectPropertyElementMaster };\r
-- OSStatus result = AudioObjectSetPropertyData( kAudioObjectSystemObject, &property, 0, NULL, sizeof(CFRunLoopRef), &theRunLoop);\r
-- if ( result != noErr ) {\r
-- errorText_ = "RtApiCore::RtApiCore: error setting run loop property!";\r
-- error( RtAudioError::WARNING );\r
-- }\r
--#endif\r
--}\r
--\r
--RtApiCore :: ~RtApiCore()\r
--{\r
-- // The subclass destructor gets called before the base class\r
-- // destructor, so close an existing stream before deallocating\r
-- // apiDeviceId memory.\r
-- if ( stream_.state != STREAM_CLOSED ) closeStream();\r
--}\r
--\r
--unsigned int RtApiCore :: getDeviceCount( void )\r
--{\r
-- // Find out how many audio devices there are, if any.\r
-- UInt32 dataSize;\r
-- AudioObjectPropertyAddress propertyAddress = { kAudioHardwarePropertyDevices, kAudioObjectPropertyScopeGlobal, kAudioObjectPropertyElementMaster };\r
-- OSStatus result = AudioObjectGetPropertyDataSize( kAudioObjectSystemObject, &propertyAddress, 0, NULL, &dataSize );\r
-- if ( result != noErr ) {\r
-- errorText_ = "RtApiCore::getDeviceCount: OS-X error getting device info!";\r
-- error( RtAudioError::WARNING );\r
-- return 0;\r
-- }\r
--\r
-- return dataSize / sizeof( AudioDeviceID );\r
--}\r
--\r
--unsigned int RtApiCore :: getDefaultInputDevice( void )\r
--{\r
-- unsigned int nDevices = getDeviceCount();\r
-- if ( nDevices <= 1 ) return 0;\r
--\r
-- AudioDeviceID id;\r
-- UInt32 dataSize = sizeof( AudioDeviceID );\r
-- AudioObjectPropertyAddress property = { kAudioHardwarePropertyDefaultInputDevice, kAudioObjectPropertyScopeGlobal, kAudioObjectPropertyElementMaster };\r
-- OSStatus result = AudioObjectGetPropertyData( kAudioObjectSystemObject, &property, 0, NULL, &dataSize, &id );\r
-- if ( result != noErr ) {\r
-- errorText_ = "RtApiCore::getDefaultInputDevice: OS-X system error getting device.";\r
-- error( RtAudioError::WARNING );\r
-- return 0;\r
-- }\r
--\r
-- dataSize *= nDevices;\r
-- AudioDeviceID deviceList[ nDevices ];\r
-- property.mSelector = kAudioHardwarePropertyDevices;\r
-- result = AudioObjectGetPropertyData( kAudioObjectSystemObject, &property, 0, NULL, &dataSize, (void *) &deviceList );\r
-- if ( result != noErr ) {\r
-- errorText_ = "RtApiCore::getDefaultInputDevice: OS-X system error getting device IDs.";\r
-- error( RtAudioError::WARNING );\r
-- return 0;\r
-- }\r
--\r
-- for ( unsigned int i=0; i<nDevices; i++ )\r
-- if ( id == deviceList[i] ) return i;\r
--\r
-- errorText_ = "RtApiCore::getDefaultInputDevice: No default device found!";\r
-- error( RtAudioError::WARNING );\r
-- return 0;\r
--}\r
--\r
--unsigned int RtApiCore :: getDefaultOutputDevice( void )\r
--{\r
-- unsigned int nDevices = getDeviceCount();\r
-- if ( nDevices <= 1 ) return 0;\r
--\r
-- AudioDeviceID id;\r
-- UInt32 dataSize = sizeof( AudioDeviceID );\r
-- AudioObjectPropertyAddress property = { kAudioHardwarePropertyDefaultOutputDevice, kAudioObjectPropertyScopeGlobal, kAudioObjectPropertyElementMaster };\r
-- OSStatus result = AudioObjectGetPropertyData( kAudioObjectSystemObject, &property, 0, NULL, &dataSize, &id );\r
-- if ( result != noErr ) {\r
-- errorText_ = "RtApiCore::getDefaultOutputDevice: OS-X system error getting device.";\r
-- error( RtAudioError::WARNING );\r
-- return 0;\r
-- }\r
--\r
-- dataSize = sizeof( AudioDeviceID ) * nDevices;\r
-- AudioDeviceID deviceList[ nDevices ];\r
-- property.mSelector = kAudioHardwarePropertyDevices;\r
-- result = AudioObjectGetPropertyData( kAudioObjectSystemObject, &property, 0, NULL, &dataSize, (void *) &deviceList );\r
-- if ( result != noErr ) {\r
-- errorText_ = "RtApiCore::getDefaultOutputDevice: OS-X system error getting device IDs.";\r
-- error( RtAudioError::WARNING );\r
-- return 0;\r
-- }\r
--\r
-- for ( unsigned int i=0; i<nDevices; i++ )\r
-- if ( id == deviceList[i] ) return i;\r
--\r
-- errorText_ = "RtApiCore::getDefaultOutputDevice: No default device found!";\r
-- error( RtAudioError::WARNING );\r
-- return 0;\r
--}\r
--\r
--RtAudio::DeviceInfo RtApiCore :: getDeviceInfo( unsigned int device )\r
--{\r
-- RtAudio::DeviceInfo info;\r
-- info.probed = false;\r
--\r
-- // Get device ID\r
-- unsigned int nDevices = getDeviceCount();\r
-- if ( nDevices == 0 ) {\r
-- errorText_ = "RtApiCore::getDeviceInfo: no devices found!";\r
-- error( RtAudioError::INVALID_USE );\r
-- return info;\r
-- }\r
--\r
-- if ( device >= nDevices ) {\r
-- errorText_ = "RtApiCore::getDeviceInfo: device ID is invalid!";\r
-- error( RtAudioError::INVALID_USE );\r
-- return info;\r
-- }\r
--\r
-- AudioDeviceID deviceList[ nDevices ];\r
-- UInt32 dataSize = sizeof( AudioDeviceID ) * nDevices;\r
-- AudioObjectPropertyAddress property = { kAudioHardwarePropertyDevices,\r
-- kAudioObjectPropertyScopeGlobal,\r
-- kAudioObjectPropertyElementMaster };\r
-- OSStatus result = AudioObjectGetPropertyData( kAudioObjectSystemObject, &property,\r
-- 0, NULL, &dataSize, (void *) &deviceList );\r
-- if ( result != noErr ) {\r
-- errorText_ = "RtApiCore::getDeviceInfo: OS-X system error getting device IDs.";\r
-- error( RtAudioError::WARNING );\r
-- return info;\r
-- }\r
--\r
-- AudioDeviceID id = deviceList[ device ];\r
--\r
-- // Get the device name.\r
-- info.name.erase();\r
-- CFStringRef cfname;\r
-- dataSize = sizeof( CFStringRef );\r
-- property.mSelector = kAudioObjectPropertyManufacturer;\r
-- result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, &cfname );\r
-- if ( result != noErr ) {\r
-- errorStream_ << "RtApiCore::probeDeviceInfo: system error (" << getErrorCode( result ) << ") getting device manufacturer.";\r
-- errorText_ = errorStream_.str();\r
-- error( RtAudioError::WARNING );\r
-- return info;\r
-- }\r
--\r
-- //const char *mname = CFStringGetCStringPtr( cfname, CFStringGetSystemEncoding() );\r
-- int length = CFStringGetLength(cfname);\r
-- char *mname = (char *)malloc(length * 3 + 1);\r
--#if defined( UNICODE ) || defined( _UNICODE )\r
-- CFStringGetCString(cfname, mname, length * 3 + 1, kCFStringEncodingUTF8);\r
--#else\r
-- CFStringGetCString(cfname, mname, length * 3 + 1, CFStringGetSystemEncoding());\r
--#endif\r
-- info.name.append( (const char *)mname, strlen(mname) );\r
-- info.name.append( ": " );\r
-- CFRelease( cfname );\r
-- free(mname);\r
--\r
-- property.mSelector = kAudioObjectPropertyName;\r
-- result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, &cfname );\r
-- if ( result != noErr ) {\r
-- errorStream_ << "RtApiCore::probeDeviceInfo: system error (" << getErrorCode( result ) << ") getting device name.";\r
-- errorText_ = errorStream_.str();\r
-- error( RtAudioError::WARNING );\r
-- return info;\r
-- }\r
--\r
-- //const char *name = CFStringGetCStringPtr( cfname, CFStringGetSystemEncoding() );\r
-- length = CFStringGetLength(cfname);\r
-- char *name = (char *)malloc(length * 3 + 1);\r
--#if defined( UNICODE ) || defined( _UNICODE )\r
-- CFStringGetCString(cfname, name, length * 3 + 1, kCFStringEncodingUTF8);\r
--#else\r
-- CFStringGetCString(cfname, name, length * 3 + 1, CFStringGetSystemEncoding());\r
--#endif\r
-- info.name.append( (const char *)name, strlen(name) );\r
-- CFRelease( cfname );\r
-- free(name);\r
--\r
-- // Get the output stream "configuration".\r
-- AudioBufferList *bufferList = nil;\r
-- property.mSelector = kAudioDevicePropertyStreamConfiguration;\r
-- property.mScope = kAudioDevicePropertyScopeOutput;\r
-- // property.mElement = kAudioObjectPropertyElementWildcard;\r
-- dataSize = 0;\r
-- result = AudioObjectGetPropertyDataSize( id, &property, 0, NULL, &dataSize );\r
-- if ( result != noErr || dataSize == 0 ) {\r
-- errorStream_ << "RtApiCore::getDeviceInfo: system error (" << getErrorCode( result ) << ") getting output stream configuration info for device (" << device << ").";\r
-- errorText_ = errorStream_.str();\r
-- error( RtAudioError::WARNING );\r
-- return info;\r
-- }\r
--\r
-- // Allocate the AudioBufferList.\r
-- bufferList = (AudioBufferList *) malloc( dataSize );\r
-- if ( bufferList == NULL ) {\r
-- errorText_ = "RtApiCore::getDeviceInfo: memory error allocating output AudioBufferList.";\r
-- error( RtAudioError::WARNING );\r
-- return info;\r
-- }\r
--\r
-- result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, bufferList );\r
-- if ( result != noErr || dataSize == 0 ) {\r
-- free( bufferList );\r
-- errorStream_ << "RtApiCore::getDeviceInfo: system error (" << getErrorCode( result ) << ") getting output stream configuration for device (" << device << ").";\r
-- errorText_ = errorStream_.str();\r
-- error( RtAudioError::WARNING );\r
-- return info;\r
-- }\r
--\r
-- // Get output channel information.\r
-- unsigned int i, nStreams = bufferList->mNumberBuffers;\r
-- for ( i=0; i<nStreams; i++ )\r
-- info.outputChannels += bufferList->mBuffers[i].mNumberChannels;\r
-- free( bufferList );\r
--\r
-- // Get the input stream "configuration".\r
-- property.mScope = kAudioDevicePropertyScopeInput;\r
-- result = AudioObjectGetPropertyDataSize( id, &property, 0, NULL, &dataSize );\r
-- if ( result != noErr || dataSize == 0 ) {\r
-- errorStream_ << "RtApiCore::getDeviceInfo: system error (" << getErrorCode( result ) << ") getting input stream configuration info for device (" << device << ").";\r
-- errorText_ = errorStream_.str();\r
-- error( RtAudioError::WARNING );\r
-- return info;\r
-- }\r
--\r
-- // Allocate the AudioBufferList.\r
-- bufferList = (AudioBufferList *) malloc( dataSize );\r
-- if ( bufferList == NULL ) {\r
-- errorText_ = "RtApiCore::getDeviceInfo: memory error allocating input AudioBufferList.";\r
-- error( RtAudioError::WARNING );\r
-- return info;\r
-- }\r
--\r
-- result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, bufferList );\r
-- if (result != noErr || dataSize == 0) {\r
-- free( bufferList );\r
-- errorStream_ << "RtApiCore::getDeviceInfo: system error (" << getErrorCode( result ) << ") getting input stream configuration for device (" << device << ").";\r
-- errorText_ = errorStream_.str();\r
-- error( RtAudioError::WARNING );\r
-- return info;\r
-- }\r
--\r
-- // Get input channel information.\r
-- nStreams = bufferList->mNumberBuffers;\r
-- for ( i=0; i<nStreams; i++ )\r
-- info.inputChannels += bufferList->mBuffers[i].mNumberChannels;\r
-- free( bufferList );\r
--\r
-- // If device opens for both playback and capture, we determine the channels.\r
-- if ( info.outputChannels > 0 && info.inputChannels > 0 )\r
-- info.duplexChannels = (info.outputChannels > info.inputChannels) ? info.inputChannels : info.outputChannels;\r
--\r
-- // Probe the device sample rates.\r
-- bool isInput = false;\r
-- if ( info.outputChannels == 0 ) isInput = true;\r
--\r
-- // Determine the supported sample rates.\r
-- property.mSelector = kAudioDevicePropertyAvailableNominalSampleRates;\r
-- if ( isInput == false ) property.mScope = kAudioDevicePropertyScopeOutput;\r
-- result = AudioObjectGetPropertyDataSize( id, &property, 0, NULL, &dataSize );\r
-- if ( result != kAudioHardwareNoError || dataSize == 0 ) {\r
-- errorStream_ << "RtApiCore::getDeviceInfo: system error (" << getErrorCode( result ) << ") getting sample rate info.";\r
-- errorText_ = errorStream_.str();\r
-- error( RtAudioError::WARNING );\r
-- return info;\r
-- }\r
--\r
-- UInt32 nRanges = dataSize / sizeof( AudioValueRange );\r
-- AudioValueRange rangeList[ nRanges ];\r
-- result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, &rangeList );\r
-- if ( result != kAudioHardwareNoError ) {\r
-- errorStream_ << "RtApiCore::getDeviceInfo: system error (" << getErrorCode( result ) << ") getting sample rates.";\r
-- errorText_ = errorStream_.str();\r
-- error( RtAudioError::WARNING );\r
-- return info;\r
-- }\r
--\r
-- // The sample rate reporting mechanism is a bit of a mystery. It\r
-- // seems that it can either return individual rates or a range of\r
-- // rates. I assume that if the min / max range values are the same,\r
-- // then that represents a single supported rate and if the min / max\r
-- // range values are different, the device supports an arbitrary\r
-- // range of values (though there might be multiple ranges, so we'll\r
-- // use the most conservative range).\r
-- Float64 minimumRate = 1.0, maximumRate = 10000000000.0;\r
-- bool haveValueRange = false;\r
-- info.sampleRates.clear();\r
-- for ( UInt32 i=0; i<nRanges; i++ ) {\r
-- if ( rangeList[i].mMinimum == rangeList[i].mMaximum ) {\r
-- unsigned int tmpSr = (unsigned int) rangeList[i].mMinimum;\r
-- info.sampleRates.push_back( tmpSr );\r
--\r
-- if ( !info.preferredSampleRate || ( tmpSr <= 48000 && tmpSr > info.preferredSampleRate ) )\r
-- info.preferredSampleRate = tmpSr;\r
--\r
-- } else {\r
-- haveValueRange = true;\r
-- if ( rangeList[i].mMinimum > minimumRate ) minimumRate = rangeList[i].mMinimum;\r
-- if ( rangeList[i].mMaximum < maximumRate ) maximumRate = rangeList[i].mMaximum;\r
-- }\r
-- }\r
--\r
-- if ( haveValueRange ) {\r
-- for ( unsigned int k=0; k<MAX_SAMPLE_RATES; k++ ) {\r
-- if ( SAMPLE_RATES[k] >= (unsigned int) minimumRate && SAMPLE_RATES[k] <= (unsigned int) maximumRate ) {\r
-- info.sampleRates.push_back( SAMPLE_RATES[k] );\r
--\r
-- if ( !info.preferredSampleRate || ( SAMPLE_RATES[k] <= 48000 && SAMPLE_RATES[k] > info.preferredSampleRate ) )\r
-- info.preferredSampleRate = SAMPLE_RATES[k];\r
-- }\r
-- }\r
-- }\r
--\r
-- // Sort and remove any redundant values\r
-- std::sort( info.sampleRates.begin(), info.sampleRates.end() );\r
-- info.sampleRates.erase( unique( info.sampleRates.begin(), info.sampleRates.end() ), info.sampleRates.end() );\r
--\r
-- if ( info.sampleRates.size() == 0 ) {\r
-- errorStream_ << "RtApiCore::probeDeviceInfo: No supported sample rates found for device (" << device << ").";\r
-- errorText_ = errorStream_.str();\r
-- error( RtAudioError::WARNING );\r
-- return info;\r
-- }\r
--\r
-- // CoreAudio always uses 32-bit floating point data for PCM streams.\r
-- // Thus, any other "physical" formats supported by the device are of\r
-- // no interest to the client.\r
-- info.nativeFormats = RTAUDIO_FLOAT32;\r
--\r
-- if ( info.outputChannels > 0 )\r
-- if ( getDefaultOutputDevice() == device ) info.isDefaultOutput = true;\r
-- if ( info.inputChannels > 0 )\r
-- if ( getDefaultInputDevice() == device ) info.isDefaultInput = true;\r
--\r
-- info.probed = true;\r
-- return info;\r
--}\r
--\r
--static OSStatus callbackHandler( AudioDeviceID inDevice,\r
-- const AudioTimeStamp* /*inNow*/,\r
-- const AudioBufferList* inInputData,\r
-- const AudioTimeStamp* /*inInputTime*/,\r
-- AudioBufferList* outOutputData,\r
-- const AudioTimeStamp* /*inOutputTime*/,\r
-- void* infoPointer )\r
--{\r
-- CallbackInfo *info = (CallbackInfo *) infoPointer;\r
--\r
-- RtApiCore *object = (RtApiCore *) info->object;\r
-- if ( object->callbackEvent( inDevice, inInputData, outOutputData ) == false )\r
-- return kAudioHardwareUnspecifiedError;\r
-- else\r
-- return kAudioHardwareNoError;\r
--}\r
--\r
--static OSStatus xrunListener( AudioObjectID /*inDevice*/,\r
-- UInt32 nAddresses,\r
-- const AudioObjectPropertyAddress properties[],\r
-- void* handlePointer )\r
--{\r
-- CoreHandle *handle = (CoreHandle *) handlePointer;\r
-- for ( UInt32 i=0; i<nAddresses; i++ ) {\r
-- if ( properties[i].mSelector == kAudioDeviceProcessorOverload ) {\r
-- if ( properties[i].mScope == kAudioDevicePropertyScopeInput )\r
-- handle->xrun[1] = true;\r
-- else\r
-- handle->xrun[0] = true;\r
-- }\r
-- }\r
--\r
-- return kAudioHardwareNoError;\r
--}\r
--\r
--static OSStatus rateListener( AudioObjectID inDevice,\r
-- UInt32 /*nAddresses*/,\r
-- const AudioObjectPropertyAddress /*properties*/[],\r
-- void* ratePointer )\r
--{\r
-- Float64 *rate = (Float64 *) ratePointer;\r
-- UInt32 dataSize = sizeof( Float64 );\r
-- AudioObjectPropertyAddress property = { kAudioDevicePropertyNominalSampleRate,\r
-- kAudioObjectPropertyScopeGlobal,\r
-- kAudioObjectPropertyElementMaster };\r
-- AudioObjectGetPropertyData( inDevice, &property, 0, NULL, &dataSize, rate );\r
-- return kAudioHardwareNoError;\r
--}\r
--\r
--bool RtApiCore :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,\r
-- unsigned int firstChannel, unsigned int sampleRate,\r
-- RtAudioFormat format, unsigned int *bufferSize,\r
-- RtAudio::StreamOptions *options )\r
--{\r
-- // Get device ID\r
-- unsigned int nDevices = getDeviceCount();\r
-- if ( nDevices == 0 ) {\r
-- // This should not happen because a check is made before this function is called.\r
-- errorText_ = "RtApiCore::probeDeviceOpen: no devices found!";\r
-- return FAILURE;\r
-- }\r
--\r
-- if ( device >= nDevices ) {\r
-- // This should not happen because a check is made before this function is called.\r
-- errorText_ = "RtApiCore::probeDeviceOpen: device ID is invalid!";\r
-- return FAILURE;\r
-- }\r
--\r
-- AudioDeviceID deviceList[ nDevices ];\r
-- UInt32 dataSize = sizeof( AudioDeviceID ) * nDevices;\r
-- AudioObjectPropertyAddress property = { kAudioHardwarePropertyDevices,\r
-- kAudioObjectPropertyScopeGlobal,\r
-- kAudioObjectPropertyElementMaster };\r
-- OSStatus result = AudioObjectGetPropertyData( kAudioObjectSystemObject, &property,\r
-- 0, NULL, &dataSize, (void *) &deviceList );\r
-- if ( result != noErr ) {\r
-- errorText_ = "RtApiCore::probeDeviceOpen: OS-X system error getting device IDs.";\r
-- return FAILURE;\r
-- }\r
--\r
-- AudioDeviceID id = deviceList[ device ];\r
--\r
-- // Setup for stream mode.\r
-- bool isInput = false;\r
-- if ( mode == INPUT ) {\r
-- isInput = true;\r
-- property.mScope = kAudioDevicePropertyScopeInput;\r
-- }\r
-- else\r
-- property.mScope = kAudioDevicePropertyScopeOutput;\r
--\r
-- // Get the stream "configuration".\r
-- AudioBufferList *bufferList = nil;\r
-- dataSize = 0;\r
-- property.mSelector = kAudioDevicePropertyStreamConfiguration;\r
-- result = AudioObjectGetPropertyDataSize( id, &property, 0, NULL, &dataSize );\r
-- if ( result != noErr || dataSize == 0 ) {\r
-- errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting stream configuration info for device (" << device << ").";\r
-- errorText_ = errorStream_.str();\r
-- return FAILURE;\r
-- }\r
--\r
-- // Allocate the AudioBufferList.\r
-- bufferList = (AudioBufferList *) malloc( dataSize );\r
-- if ( bufferList == NULL ) {\r
-- errorText_ = "RtApiCore::probeDeviceOpen: memory error allocating AudioBufferList.";\r
-- return FAILURE;\r
-- }\r
--\r
-- result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, bufferList );\r
-- if (result != noErr || dataSize == 0) {\r
-- free( bufferList );\r
-- errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting stream configuration for device (" << device << ").";\r
-- errorText_ = errorStream_.str();\r
-- return FAILURE;\r
-- }\r
--\r
-- // Search for one or more streams that contain the desired number of\r
-- // channels. CoreAudio devices can have an arbitrary number of\r
-- // streams and each stream can have an arbitrary number of channels.\r
-- // For each stream, a single buffer of interleaved samples is\r
-- // provided. RtAudio prefers the use of one stream of interleaved\r
-- // data or multiple consecutive single-channel streams. However, we\r
-- // now support multiple consecutive multi-channel streams of\r
-- // interleaved data as well.\r
-- UInt32 iStream, offsetCounter = firstChannel;\r
-- UInt32 nStreams = bufferList->mNumberBuffers;\r
-- bool monoMode = false;\r
-- bool foundStream = false;\r
--\r
-- // First check that the device supports the requested number of\r
-- // channels.\r
-- UInt32 deviceChannels = 0;\r
-- for ( iStream=0; iStream<nStreams; iStream++ )\r
-- deviceChannels += bufferList->mBuffers[iStream].mNumberChannels;\r
--\r
-- if ( deviceChannels < ( channels + firstChannel ) ) {\r
-- free( bufferList );\r
-- errorStream_ << "RtApiCore::probeDeviceOpen: the device (" << device << ") does not support the requested channel count.";\r
-- errorText_ = errorStream_.str();\r
-- return FAILURE;\r
-- }\r
--\r
-- // Look for a single stream meeting our needs.\r
-- UInt32 firstStream, streamCount = 1, streamChannels = 0, channelOffset = 0;\r
-- for ( iStream=0; iStream<nStreams; iStream++ ) {\r
-- streamChannels = bufferList->mBuffers[iStream].mNumberChannels;\r
-- if ( streamChannels >= channels + offsetCounter ) {\r
-- firstStream = iStream;\r
-- channelOffset = offsetCounter;\r
-- foundStream = true;\r
-- break;\r
-- }\r
-- if ( streamChannels > offsetCounter ) break;\r
-- offsetCounter -= streamChannels;\r
-- }\r
--\r
-- // If we didn't find a single stream above, then we should be able\r
-- // to meet the channel specification with multiple streams.\r
-- if ( foundStream == false ) {\r
-- monoMode = true;\r
-- offsetCounter = firstChannel;\r
-- for ( iStream=0; iStream<nStreams; iStream++ ) {\r
-- streamChannels = bufferList->mBuffers[iStream].mNumberChannels;\r
-- if ( streamChannels > offsetCounter ) break;\r
-- offsetCounter -= streamChannels;\r
-- }\r
--\r
-- firstStream = iStream;\r
-- channelOffset = offsetCounter;\r
-- Int32 channelCounter = channels + offsetCounter - streamChannels;\r
--\r
-- if ( streamChannels > 1 ) monoMode = false;\r
-- while ( channelCounter > 0 ) {\r
-- streamChannels = bufferList->mBuffers[++iStream].mNumberChannels;\r
-- if ( streamChannels > 1 ) monoMode = false;\r
-- channelCounter -= streamChannels;\r
-- streamCount++;\r
-- }\r
-- }\r
--\r
-- free( bufferList );\r
--\r
-- // Determine the buffer size.\r
-- AudioValueRange bufferRange;\r
-- dataSize = sizeof( AudioValueRange );\r
-- property.mSelector = kAudioDevicePropertyBufferFrameSizeRange;\r
-- result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, &bufferRange );\r
--\r
-- if ( result != noErr ) {\r
-- errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting buffer size range for device (" << device << ").";\r
-- errorText_ = errorStream_.str();\r
-- return FAILURE;\r
-- }\r
--\r
-- if ( bufferRange.mMinimum > *bufferSize ) *bufferSize = (unsigned long) bufferRange.mMinimum;\r
-- else if ( bufferRange.mMaximum < *bufferSize ) *bufferSize = (unsigned long) bufferRange.mMaximum;\r
-- if ( options && options->flags & RTAUDIO_MINIMIZE_LATENCY ) *bufferSize = (unsigned long) bufferRange.mMinimum;\r
--\r
-- // Set the buffer size. For multiple streams, I'm assuming we only\r
-- // need to make this setting for the master channel.\r
-- UInt32 theSize = (UInt32) *bufferSize;\r
-- dataSize = sizeof( UInt32 );\r
-- property.mSelector = kAudioDevicePropertyBufferFrameSize;\r
-- result = AudioObjectSetPropertyData( id, &property, 0, NULL, dataSize, &theSize );\r
--\r
-- if ( result != noErr ) {\r
-- errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") setting the buffer size for device (" << device << ").";\r
-- errorText_ = errorStream_.str();\r
-- return FAILURE;\r
-- }\r
--\r
-- // If attempting to setup a duplex stream, the bufferSize parameter\r
-- // MUST be the same in both directions!\r
-- *bufferSize = theSize;\r
-- if ( stream_.mode == OUTPUT && mode == INPUT && *bufferSize != stream_.bufferSize ) {\r
-- errorStream_ << "RtApiCore::probeDeviceOpen: system error setting buffer size for duplex stream on device (" << device << ").";\r
-- errorText_ = errorStream_.str();\r
-- return FAILURE;\r
-- }\r
--\r
-- stream_.bufferSize = *bufferSize;\r
-- stream_.nBuffers = 1;\r
--\r
-- // Try to set "hog" mode ... it's not clear to me this is working.\r
-- if ( options && options->flags & RTAUDIO_HOG_DEVICE ) {\r
-- pid_t hog_pid;\r
-- dataSize = sizeof( hog_pid );\r
-- property.mSelector = kAudioDevicePropertyHogMode;\r
-- result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, &hog_pid );\r
-- if ( result != noErr ) {\r
-- errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting 'hog' state!";\r
-- errorText_ = errorStream_.str();\r
-- return FAILURE;\r
-- }\r
--\r
-- if ( hog_pid != getpid() ) {\r
-- hog_pid = getpid();\r
-- result = AudioObjectSetPropertyData( id, &property, 0, NULL, dataSize, &hog_pid );\r
-- if ( result != noErr ) {\r
-- errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") setting 'hog' state!";\r
-- errorText_ = errorStream_.str();\r
-- return FAILURE;\r
-- }\r
-- }\r
-- }\r
--\r
-- // Check and if necessary, change the sample rate for the device.\r
-- Float64 nominalRate;\r
-- dataSize = sizeof( Float64 );\r
-- property.mSelector = kAudioDevicePropertyNominalSampleRate;\r
-- result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, &nominalRate );\r
-- if ( result != noErr ) {\r
-- errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting current sample rate.";\r
-- errorText_ = errorStream_.str();\r
-- return FAILURE;\r
-- }\r
--\r
-- // Only change the sample rate if off by more than 1 Hz.\r
-- if ( fabs( nominalRate - (double)sampleRate ) > 1.0 ) {\r
--\r
-- // Set a property listener for the sample rate change\r
-- Float64 reportedRate = 0.0;\r
-- AudioObjectPropertyAddress tmp = { kAudioDevicePropertyNominalSampleRate, kAudioObjectPropertyScopeGlobal, kAudioObjectPropertyElementMaster };\r
-- result = AudioObjectAddPropertyListener( id, &tmp, rateListener, (void *) &reportedRate );\r
-- if ( result != noErr ) {\r
-- errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") setting sample rate property listener for device (" << device << ").";\r
-- errorText_ = errorStream_.str();\r
-- return FAILURE;\r
-- }\r
--\r
-- nominalRate = (Float64) sampleRate;\r
-- result = AudioObjectSetPropertyData( id, &property, 0, NULL, dataSize, &nominalRate );\r
-- if ( result != noErr ) {\r
-- AudioObjectRemovePropertyListener( id, &tmp, rateListener, (void *) &reportedRate );\r
-- errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") setting sample rate for device (" << device << ").";\r
-- errorText_ = errorStream_.str();\r
-- return FAILURE;\r
-- }\r
--\r
-- // Now wait until the reported nominal rate is what we just set.\r
-- UInt32 microCounter = 0;\r
-- while ( reportedRate != nominalRate ) {\r
-- microCounter += 5000;\r
-- if ( microCounter > 5000000 ) break;\r
-- usleep( 5000 );\r
-- }\r
--\r
-- // Remove the property listener.\r
-- AudioObjectRemovePropertyListener( id, &tmp, rateListener, (void *) &reportedRate );\r
--\r
-- if ( microCounter > 5000000 ) {\r
-- errorStream_ << "RtApiCore::probeDeviceOpen: timeout waiting for sample rate update for device (" << device << ").";\r
-- errorText_ = errorStream_.str();\r
-- return FAILURE;\r
-- }\r
-- }\r
--\r
-- // Now set the stream format for all streams. Also, check the\r
-- // physical format of the device and change that if necessary.\r
-- AudioStreamBasicDescription description;\r
-- dataSize = sizeof( AudioStreamBasicDescription );\r
-- property.mSelector = kAudioStreamPropertyVirtualFormat;\r
-- result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, &description );\r
-- if ( result != noErr ) {\r
-- errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting stream format for device (" << device << ").";\r
-- errorText_ = errorStream_.str();\r
-- return FAILURE;\r
-- }\r
--\r
-- // Set the sample rate and data format id. However, only make the\r
-- // change if the sample rate is not within 1.0 of the desired\r
-- // rate and the format is not linear pcm.\r
-- bool updateFormat = false;\r
-- if ( fabs( description.mSampleRate - (Float64)sampleRate ) > 1.0 ) {\r
-- description.mSampleRate = (Float64) sampleRate;\r
-- updateFormat = true;\r
-- }\r
--\r
-- if ( description.mFormatID != kAudioFormatLinearPCM ) {\r
-- description.mFormatID = kAudioFormatLinearPCM;\r
-- updateFormat = true;\r
-- }\r
--\r
-- if ( updateFormat ) {\r
-- result = AudioObjectSetPropertyData( id, &property, 0, NULL, dataSize, &description );\r
-- if ( result != noErr ) {\r
-- errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") setting sample rate or data format for device (" << device << ").";\r
-- errorText_ = errorStream_.str();\r
-- return FAILURE;\r
-- }\r
-- }\r
--\r
-- // Now check the physical format.\r
-- property.mSelector = kAudioStreamPropertyPhysicalFormat;\r
-- result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, &description );\r
-- if ( result != noErr ) {\r
-- errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting stream physical format for device (" << device << ").";\r
-- errorText_ = errorStream_.str();\r
-- return FAILURE;\r
-- }\r
--\r
-- //std::cout << "Current physical stream format:" << std::endl;\r
-- //std::cout << " mBitsPerChan = " << description.mBitsPerChannel << std::endl;\r
-- //std::cout << " aligned high = " << (description.mFormatFlags & kAudioFormatFlagIsAlignedHigh) << ", isPacked = " << (description.mFormatFlags & kAudioFormatFlagIsPacked) << std::endl;\r
-- //std::cout << " bytesPerFrame = " << description.mBytesPerFrame << std::endl;\r
-- //std::cout << " sample rate = " << description.mSampleRate << std::endl;\r
--\r
-- if ( description.mFormatID != kAudioFormatLinearPCM || description.mBitsPerChannel < 16 ) {\r
-- description.mFormatID = kAudioFormatLinearPCM;\r
-- //description.mSampleRate = (Float64) sampleRate;\r
-- AudioStreamBasicDescription testDescription = description;\r
-- UInt32 formatFlags;\r
--\r
-- // We'll try higher bit rates first and then work our way down.\r
-- std::vector< std::pair<UInt32, UInt32> > physicalFormats;\r
-- formatFlags = (description.mFormatFlags | kLinearPCMFormatFlagIsFloat) & ~kLinearPCMFormatFlagIsSignedInteger;\r
-- physicalFormats.push_back( std::pair<Float32, UInt32>( 32, formatFlags ) );\r
-- formatFlags = (description.mFormatFlags | kLinearPCMFormatFlagIsSignedInteger | kAudioFormatFlagIsPacked) & ~kLinearPCMFormatFlagIsFloat;\r
-- physicalFormats.push_back( std::pair<Float32, UInt32>( 32, formatFlags ) );\r
-- physicalFormats.push_back( std::pair<Float32, UInt32>( 24, formatFlags ) ); // 24-bit packed\r
-- formatFlags &= ~( kAudioFormatFlagIsPacked | kAudioFormatFlagIsAlignedHigh );\r
-- physicalFormats.push_back( std::pair<Float32, UInt32>( 24.2, formatFlags ) ); // 24-bit in 4 bytes, aligned low\r
-- formatFlags |= kAudioFormatFlagIsAlignedHigh;\r
-- physicalFormats.push_back( std::pair<Float32, UInt32>( 24.4, formatFlags ) ); // 24-bit in 4 bytes, aligned high\r
-- formatFlags = (description.mFormatFlags | kLinearPCMFormatFlagIsSignedInteger | kAudioFormatFlagIsPacked) & ~kLinearPCMFormatFlagIsFloat;\r
-- physicalFormats.push_back( std::pair<Float32, UInt32>( 16, formatFlags ) );\r
-- physicalFormats.push_back( std::pair<Float32, UInt32>( 8, formatFlags ) );\r
--\r
-- bool setPhysicalFormat = false;\r
-- for( unsigned int i=0; i<physicalFormats.size(); i++ ) {\r
-- testDescription = description;\r
-- testDescription.mBitsPerChannel = (UInt32) physicalFormats[i].first;\r
-- testDescription.mFormatFlags = physicalFormats[i].second;\r
-- if ( (24 == (UInt32)physicalFormats[i].first) && ~( physicalFormats[i].second & kAudioFormatFlagIsPacked ) )\r
-- testDescription.mBytesPerFrame = 4 * testDescription.mChannelsPerFrame;\r
-- else\r
-- testDescription.mBytesPerFrame = testDescription.mBitsPerChannel/8 * testDescription.mChannelsPerFrame;\r
-- testDescription.mBytesPerPacket = testDescription.mBytesPerFrame * testDescription.mFramesPerPacket;\r
-- result = AudioObjectSetPropertyData( id, &property, 0, NULL, dataSize, &testDescription );\r
-- if ( result == noErr ) {\r
-- setPhysicalFormat = true;\r
-- //std::cout << "Updated physical stream format:" << std::endl;\r
-- //std::cout << " mBitsPerChan = " << testDescription.mBitsPerChannel << std::endl;\r
-- //std::cout << " aligned high = " << (testDescription.mFormatFlags & kAudioFormatFlagIsAlignedHigh) << ", isPacked = " << (testDescription.mFormatFlags & kAudioFormatFlagIsPacked) << std::endl;\r
-- //std::cout << " bytesPerFrame = " << testDescription.mBytesPerFrame << std::endl;\r
-- //std::cout << " sample rate = " << testDescription.mSampleRate << std::endl;\r
-- break;\r
-- }\r
-- }\r
--\r
-- if ( !setPhysicalFormat ) {\r
-- errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") setting physical data format for device (" << device << ").";\r
-- errorText_ = errorStream_.str();\r
-- return FAILURE;\r
-- }\r
-- } // done setting virtual/physical formats.\r
--\r
-- // Get the stream / device latency.\r
-- UInt32 latency;\r
-- dataSize = sizeof( UInt32 );\r
-- property.mSelector = kAudioDevicePropertyLatency;\r
-- if ( AudioObjectHasProperty( id, &property ) == true ) {\r
-- result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, &latency );\r
-- if ( result == kAudioHardwareNoError ) stream_.latency[ mode ] = latency;\r
-- else {\r
-- errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting device latency for device (" << device << ").";\r
-- errorText_ = errorStream_.str();\r
-- error( RtAudioError::WARNING );\r
-- }\r
-- }\r
--\r
-- // Byte-swapping: According to AudioHardware.h, the stream data will\r
-- // always be presented in native-endian format, so we should never\r
-- // need to byte swap.\r
-- stream_.doByteSwap[mode] = false;\r
--\r
-- // From the CoreAudio documentation, PCM data must be supplied as\r
-- // 32-bit floats.\r
-- stream_.userFormat = format;\r
-- stream_.deviceFormat[mode] = RTAUDIO_FLOAT32;\r
--\r
-- if ( streamCount == 1 )\r
-- stream_.nDeviceChannels[mode] = description.mChannelsPerFrame;\r
-- else // multiple streams\r
-- stream_.nDeviceChannels[mode] = channels;\r
-- stream_.nUserChannels[mode] = channels;\r
-- stream_.channelOffset[mode] = channelOffset; // offset within a CoreAudio stream\r
-- if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) stream_.userInterleaved = false;\r
-- else stream_.userInterleaved = true;\r
-- stream_.deviceInterleaved[mode] = true;\r
-- if ( monoMode == true ) stream_.deviceInterleaved[mode] = false;\r
--\r
-- // Set flags for buffer conversion.\r
-- stream_.doConvertBuffer[mode] = false;\r
-- if ( stream_.userFormat != stream_.deviceFormat[mode] )\r
-- stream_.doConvertBuffer[mode] = true;\r
-- if ( stream_.nUserChannels[mode] < stream_.nDeviceChannels[mode] )\r
-- stream_.doConvertBuffer[mode] = true;\r
-- if ( streamCount == 1 ) {\r
-- if ( stream_.nUserChannels[mode] > 1 &&\r
-- stream_.userInterleaved != stream_.deviceInterleaved[mode] )\r
-- stream_.doConvertBuffer[mode] = true;\r
-- }\r
-- else if ( monoMode && stream_.userInterleaved )\r
-- stream_.doConvertBuffer[mode] = true;\r
--\r
-- // Allocate our CoreHandle structure for the stream.\r
-- CoreHandle *handle = 0;\r
-- if ( stream_.apiHandle == 0 ) {\r
-- try {\r
-- handle = new CoreHandle;\r
-- }\r
-- catch ( std::bad_alloc& ) {\r
-- errorText_ = "RtApiCore::probeDeviceOpen: error allocating CoreHandle memory.";\r
-- goto error;\r
-- }\r
--\r
-- if ( pthread_cond_init( &handle->condition, NULL ) ) {\r
-- errorText_ = "RtApiCore::probeDeviceOpen: error initializing pthread condition variable.";\r
-- goto error;\r
-- }\r
-- stream_.apiHandle = (void *) handle;\r
-- }\r
-- else\r
-- handle = (CoreHandle *) stream_.apiHandle;\r
-- handle->iStream[mode] = firstStream;\r
-- handle->nStreams[mode] = streamCount;\r
-- handle->id[mode] = id;\r
--\r
-- // Allocate necessary internal buffers.\r
-- unsigned long bufferBytes;\r
-- bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat );\r
-- // stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 );\r
-- stream_.userBuffer[mode] = (char *) malloc( bufferBytes * sizeof(char) );\r
-- memset( stream_.userBuffer[mode], 0, bufferBytes * sizeof(char) );\r
-- if ( stream_.userBuffer[mode] == NULL ) {\r
-- errorText_ = "RtApiCore::probeDeviceOpen: error allocating user buffer memory.";\r
-- goto error;\r
-- }\r
--\r
-- // If possible, we will make use of the CoreAudio stream buffers as\r
-- // "device buffers". However, we can't do this if using multiple\r
-- // streams.\r
-- if ( stream_.doConvertBuffer[mode] && handle->nStreams[mode] > 1 ) {\r
--\r
-- bool makeBuffer = true;\r
-- bufferBytes = stream_.nDeviceChannels[mode] * formatBytes( stream_.deviceFormat[mode] );\r
-- if ( mode == INPUT ) {\r
-- if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) {\r
-- unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] );\r
-- if ( bufferBytes <= bytesOut ) makeBuffer = false;\r
-- }\r
-- }\r
--\r
-- if ( makeBuffer ) {\r
-- bufferBytes *= *bufferSize;\r
-- if ( stream_.deviceBuffer ) free( stream_.deviceBuffer );\r
-- stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 );\r
-- if ( stream_.deviceBuffer == NULL ) {\r
-- errorText_ = "RtApiCore::probeDeviceOpen: error allocating device buffer memory.";\r
-- goto error;\r
-- }\r
-- }\r
-- }\r
--\r
-- stream_.sampleRate = sampleRate;\r
-- stream_.device[mode] = device;\r
-- stream_.state = STREAM_STOPPED;\r
-- stream_.callbackInfo.object = (void *) this;\r
--\r
-- // Setup the buffer conversion information structure.\r
-- if ( stream_.doConvertBuffer[mode] ) {\r
-- if ( streamCount > 1 ) setConvertInfo( mode, 0 );\r
-- else setConvertInfo( mode, channelOffset );\r
-- }\r
--\r
-- if ( mode == INPUT && stream_.mode == OUTPUT && stream_.device[0] == device )\r
-- // Only one callback procedure per device.\r
-- stream_.mode = DUPLEX;\r
-- else {\r
--#if defined( MAC_OS_X_VERSION_10_5 ) && ( MAC_OS_X_VERSION_MIN_REQUIRED >= MAC_OS_X_VERSION_10_5 )\r
-- result = AudioDeviceCreateIOProcID( id, callbackHandler, (void *) &stream_.callbackInfo, &handle->procId[mode] );\r
--#else\r
-- // deprecated in favor of AudioDeviceCreateIOProcID()\r
-- result = AudioDeviceAddIOProc( id, callbackHandler, (void *) &stream_.callbackInfo );\r
--#endif\r
-- if ( result != noErr ) {\r
-- errorStream_ << "RtApiCore::probeDeviceOpen: system error setting callback for device (" << device << ").";\r
-- errorText_ = errorStream_.str();\r
-- goto error;\r
-- }\r
-- if ( stream_.mode == OUTPUT && mode == INPUT )\r
-- stream_.mode = DUPLEX;\r
-- else\r
-- stream_.mode = mode;\r
-- }\r
--\r
-- // Setup the device property listener for over/underload.\r
-- property.mSelector = kAudioDeviceProcessorOverload;\r
-- property.mScope = kAudioObjectPropertyScopeGlobal;\r
-- result = AudioObjectAddPropertyListener( id, &property, xrunListener, (void *) handle );\r
--\r
-- return SUCCESS;\r
--\r
-- error:\r
-- if ( handle ) {\r
-- pthread_cond_destroy( &handle->condition );\r
-- delete handle;\r
-- stream_.apiHandle = 0;\r
-- }\r
--\r
-- for ( int i=0; i<2; i++ ) {\r
-- if ( stream_.userBuffer[i] ) {\r
-- free( stream_.userBuffer[i] );\r
-- stream_.userBuffer[i] = 0;\r
-- }\r
-- }\r
--\r
-- if ( stream_.deviceBuffer ) {\r
-- free( stream_.deviceBuffer );\r
-- stream_.deviceBuffer = 0;\r
-- }\r
--\r
-- stream_.state = STREAM_CLOSED;\r
-- return FAILURE;\r
--}\r
--\r
--void RtApiCore :: closeStream( void )\r
--{\r
-- if ( stream_.state == STREAM_CLOSED ) {\r
-- errorText_ = "RtApiCore::closeStream(): no open stream to close!";\r
-- error( RtAudioError::WARNING );\r
-- return;\r
-- }\r
--\r
-- CoreHandle *handle = (CoreHandle *) stream_.apiHandle;\r
-- if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {\r
-- if (handle) {\r
-- AudioObjectPropertyAddress property = { kAudioHardwarePropertyDevices,\r
-- kAudioObjectPropertyScopeGlobal,\r
-- kAudioObjectPropertyElementMaster };\r
--\r
-- property.mSelector = kAudioDeviceProcessorOverload;\r
-- property.mScope = kAudioObjectPropertyScopeGlobal;\r
-- if (AudioObjectRemovePropertyListener( handle->id[0], &property, xrunListener, (void *) handle ) != noErr) {\r
-- errorText_ = "RtApiCore::closeStream(): error removing property listener!";\r
-- error( RtAudioError::WARNING );\r
-- }\r
-- }\r
-- if ( stream_.state == STREAM_RUNNING )\r
-- AudioDeviceStop( handle->id[0], callbackHandler );\r
--#if defined( MAC_OS_X_VERSION_10_5 ) && ( MAC_OS_X_VERSION_MIN_REQUIRED >= MAC_OS_X_VERSION_10_5 )\r
-- AudioDeviceDestroyIOProcID( handle->id[0], handle->procId[0] );\r
--#else\r
-- // deprecated in favor of AudioDeviceDestroyIOProcID()\r
-- AudioDeviceRemoveIOProc( handle->id[0], callbackHandler );\r
--#endif\r
-- }\r
--\r
-- if ( stream_.mode == INPUT || ( stream_.mode == DUPLEX && stream_.device[0] != stream_.device[1] ) ) {\r
-- if (handle) {\r
-- AudioObjectPropertyAddress property = { kAudioHardwarePropertyDevices,\r
-- kAudioObjectPropertyScopeGlobal,\r
-- kAudioObjectPropertyElementMaster };\r
--\r
-- property.mSelector = kAudioDeviceProcessorOverload;\r
-- property.mScope = kAudioObjectPropertyScopeGlobal;\r
-- if (AudioObjectRemovePropertyListener( handle->id[1], &property, xrunListener, (void *) handle ) != noErr) {\r
-- errorText_ = "RtApiCore::closeStream(): error removing property listener!";\r
-- error( RtAudioError::WARNING );\r
-- }\r
-- }\r
-- if ( stream_.state == STREAM_RUNNING )\r
-- AudioDeviceStop( handle->id[1], callbackHandler );\r
--#if defined( MAC_OS_X_VERSION_10_5 ) && ( MAC_OS_X_VERSION_MIN_REQUIRED >= MAC_OS_X_VERSION_10_5 )\r
-- AudioDeviceDestroyIOProcID( handle->id[1], handle->procId[1] );\r
--#else\r
-- // deprecated in favor of AudioDeviceDestroyIOProcID()\r
-- AudioDeviceRemoveIOProc( handle->id[1], callbackHandler );\r
--#endif\r
-- }\r
--\r
-- for ( int i=0; i<2; i++ ) {\r
-- if ( stream_.userBuffer[i] ) {\r
-- free( stream_.userBuffer[i] );\r
-- stream_.userBuffer[i] = 0;\r
-- }\r
-- }\r
--\r
-- if ( stream_.deviceBuffer ) {\r
-- free( stream_.deviceBuffer );\r
-- stream_.deviceBuffer = 0;\r
-- }\r
--\r
-- // Destroy pthread condition variable.\r
-- pthread_cond_destroy( &handle->condition );\r
-- delete handle;\r
-- stream_.apiHandle = 0;\r
--\r
-- stream_.mode = UNINITIALIZED;\r
-- stream_.state = STREAM_CLOSED;\r
--}\r
--\r
--void RtApiCore :: startStream( void )\r
--{\r
-- verifyStream();\r
-- if ( stream_.state == STREAM_RUNNING ) {\r
-- errorText_ = "RtApiCore::startStream(): the stream is already running!";\r
-- error( RtAudioError::WARNING );\r
-- return;\r
-- }\r
--\r
-- OSStatus result = noErr;\r
-- CoreHandle *handle = (CoreHandle *) stream_.apiHandle;\r
-- if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {\r
--\r
-- result = AudioDeviceStart( handle->id[0], callbackHandler );\r
-- if ( result != noErr ) {\r
-- errorStream_ << "RtApiCore::startStream: system error (" << getErrorCode( result ) << ") starting callback procedure on device (" << stream_.device[0] << ").";\r
-- errorText_ = errorStream_.str();\r
-- goto unlock;\r
-- }\r
-- }\r
--\r
-- if ( stream_.mode == INPUT ||\r
-- ( stream_.mode == DUPLEX && stream_.device[0] != stream_.device[1] ) ) {\r
--\r
-- result = AudioDeviceStart( handle->id[1], callbackHandler );\r
-- if ( result != noErr ) {\r
-- errorStream_ << "RtApiCore::startStream: system error starting input callback procedure on device (" << stream_.device[1] << ").";\r
-- errorText_ = errorStream_.str();\r
-- goto unlock;\r
-- }\r
-- }\r
--\r
-- handle->drainCounter = 0;\r
-- handle->internalDrain = false;\r
-- stream_.state = STREAM_RUNNING;\r
--\r
-- unlock:\r
-- if ( result == noErr ) return;\r
-- error( RtAudioError::SYSTEM_ERROR );\r
--}\r
--\r
--void RtApiCore :: stopStream( void )\r
--{\r
-- verifyStream();\r
-- if ( stream_.state == STREAM_STOPPED ) {\r
-- errorText_ = "RtApiCore::stopStream(): the stream is already stopped!";\r
-- error( RtAudioError::WARNING );\r
-- return;\r
-- }\r
--\r
-- OSStatus result = noErr;\r
-- CoreHandle *handle = (CoreHandle *) stream_.apiHandle;\r
-- if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {\r
--\r
-- if ( handle->drainCounter == 0 ) {\r
-- handle->drainCounter = 2;\r
-- pthread_cond_wait( &handle->condition, &stream_.mutex ); // block until signaled\r
-- }\r
--\r
-- result = AudioDeviceStop( handle->id[0], callbackHandler );\r
-- if ( result != noErr ) {\r
-- errorStream_ << "RtApiCore::stopStream: system error (" << getErrorCode( result ) << ") stopping callback procedure on device (" << stream_.device[0] << ").";\r
-- errorText_ = errorStream_.str();\r
-- goto unlock;\r
-- }\r
-- }\r
--\r
-- if ( stream_.mode == INPUT || ( stream_.mode == DUPLEX && stream_.device[0] != stream_.device[1] ) ) {\r
--\r
-- result = AudioDeviceStop( handle->id[1], callbackHandler );\r
-- if ( result != noErr ) {\r
-- errorStream_ << "RtApiCore::stopStream: system error (" << getErrorCode( result ) << ") stopping input callback procedure on device (" << stream_.device[1] << ").";\r
-- errorText_ = errorStream_.str();\r
-- goto unlock;\r
-- }\r
-- }\r
--\r
-- stream_.state = STREAM_STOPPED;\r
--\r
-- unlock:\r
-- if ( result == noErr ) return;\r
-- error( RtAudioError::SYSTEM_ERROR );\r
--}\r
--\r
--void RtApiCore :: abortStream( void )\r
--{\r
-- verifyStream();\r
-- if ( stream_.state == STREAM_STOPPED ) {\r
-- errorText_ = "RtApiCore::abortStream(): the stream is already stopped!";\r
-- error( RtAudioError::WARNING );\r
-- return;\r
-- }\r
--\r
-- CoreHandle *handle = (CoreHandle *) stream_.apiHandle;\r
-- handle->drainCounter = 2;\r
--\r
-- stopStream();\r
--}\r
--\r
--// This function will be called by a spawned thread when the user\r
--// callback function signals that the stream should be stopped or\r
--// aborted. It is better to handle it this way because the\r
--// callbackEvent() function probably should return before the AudioDeviceStop()\r
--// function is called.\r
--static void *coreStopStream( void *ptr )\r
--{\r
-- CallbackInfo *info = (CallbackInfo *) ptr;\r
-- RtApiCore *object = (RtApiCore *) info->object;\r
--\r
-- object->stopStream();\r
-- pthread_exit( NULL );\r
--}\r
--\r
--bool RtApiCore :: callbackEvent( AudioDeviceID deviceId,\r
-- const AudioBufferList *inBufferList,\r
-- const AudioBufferList *outBufferList )\r
--{\r
-- if ( stream_.state == STREAM_STOPPED || stream_.state == STREAM_STOPPING ) return SUCCESS;\r
-- if ( stream_.state == STREAM_CLOSED ) {\r
-- errorText_ = "RtApiCore::callbackEvent(): the stream is closed ... this shouldn't happen!";\r
-- error( RtAudioError::WARNING );\r
-- return FAILURE;\r
-- }\r
--\r
-- CallbackInfo *info = (CallbackInfo *) &stream_.callbackInfo;\r
-- CoreHandle *handle = (CoreHandle *) stream_.apiHandle;\r
--\r
-- // Check if we were draining the stream and signal is finished.\r
-- if ( handle->drainCounter > 3 ) {\r
-- ThreadHandle threadId;\r
--\r
-- stream_.state = STREAM_STOPPING;\r
-- if ( handle->internalDrain == true )\r
-- pthread_create( &threadId, NULL, coreStopStream, info );\r
-- else // external call to stopStream()\r
-- pthread_cond_signal( &handle->condition );\r
-- return SUCCESS;\r
-- }\r
--\r
-- AudioDeviceID outputDevice = handle->id[0];\r
--\r
-- // Invoke user callback to get fresh output data UNLESS we are\r
-- // draining stream or duplex mode AND the input/output devices are\r
-- // different AND this function is called for the input device.\r
-- if ( handle->drainCounter == 0 && ( stream_.mode != DUPLEX || deviceId == outputDevice ) ) {\r
-- RtAudioCallback callback = (RtAudioCallback) info->callback;\r
-- double streamTime = getStreamTime();\r
-- RtAudioStreamStatus status = 0;\r
-- if ( stream_.mode != INPUT && handle->xrun[0] == true ) {\r
-- status |= RTAUDIO_OUTPUT_UNDERFLOW;\r
-- handle->xrun[0] = false;\r
-- }\r
-- if ( stream_.mode != OUTPUT && handle->xrun[1] == true ) {\r
-- status |= RTAUDIO_INPUT_OVERFLOW;\r
-- handle->xrun[1] = false;\r
-- }\r
--\r
-- int cbReturnValue = callback( stream_.userBuffer[0], stream_.userBuffer[1],\r
-- stream_.bufferSize, streamTime, status, info->userData );\r
-- if ( cbReturnValue == 2 ) {\r
-- stream_.state = STREAM_STOPPING;\r
-- handle->drainCounter = 2;\r
-- abortStream();\r
-- return SUCCESS;\r
-- }\r
-- else if ( cbReturnValue == 1 ) {\r
-- handle->drainCounter = 1;\r
-- handle->internalDrain = true;\r
-- }\r
-- }\r
--\r
-- if ( stream_.mode == OUTPUT || ( stream_.mode == DUPLEX && deviceId == outputDevice ) ) {\r
--\r
-- if ( handle->drainCounter > 1 ) { // write zeros to the output stream\r
--\r
-- if ( handle->nStreams[0] == 1 ) {\r
-- memset( outBufferList->mBuffers[handle->iStream[0]].mData,\r
-- 0,\r
-- outBufferList->mBuffers[handle->iStream[0]].mDataByteSize );\r
-- }\r
-- else { // fill multiple streams with zeros\r
-- for ( unsigned int i=0; i<handle->nStreams[0]; i++ ) {\r
-- memset( outBufferList->mBuffers[handle->iStream[0]+i].mData,\r
-- 0,\r
-- outBufferList->mBuffers[handle->iStream[0]+i].mDataByteSize );\r
-- }\r
-- }\r
-- }\r
-- else if ( handle->nStreams[0] == 1 ) {\r
-- if ( stream_.doConvertBuffer[0] ) { // convert directly to CoreAudio stream buffer\r
-- convertBuffer( (char *) outBufferList->mBuffers[handle->iStream[0]].mData,\r
-- stream_.userBuffer[0], stream_.convertInfo[0] );\r
-- }\r
-- else { // copy from user buffer\r
-- memcpy( outBufferList->mBuffers[handle->iStream[0]].mData,\r
-- stream_.userBuffer[0],\r
-- outBufferList->mBuffers[handle->iStream[0]].mDataByteSize );\r
-- }\r
-- }\r
-- else { // fill multiple streams\r
-- Float32 *inBuffer = (Float32 *) stream_.userBuffer[0];\r
-- if ( stream_.doConvertBuffer[0] ) {\r
-- convertBuffer( stream_.deviceBuffer, stream_.userBuffer[0], stream_.convertInfo[0] );\r
-- inBuffer = (Float32 *) stream_.deviceBuffer;\r
-- }\r
--\r
-- if ( stream_.deviceInterleaved[0] == false ) { // mono mode\r
-- UInt32 bufferBytes = outBufferList->mBuffers[handle->iStream[0]].mDataByteSize;\r
-- for ( unsigned int i=0; i<stream_.nUserChannels[0]; i++ ) {\r
-- memcpy( outBufferList->mBuffers[handle->iStream[0]+i].mData,\r
-- (void *)&inBuffer[i*stream_.bufferSize], bufferBytes );\r
-- }\r
-- }\r
-- else { // fill multiple multi-channel streams with interleaved data\r
-- UInt32 streamChannels, channelsLeft, inJump, outJump, inOffset;\r
-- Float32 *out, *in;\r
--\r
-- bool inInterleaved = ( stream_.userInterleaved ) ? true : false;\r
-- UInt32 inChannels = stream_.nUserChannels[0];\r
-- if ( stream_.doConvertBuffer[0] ) {\r
-- inInterleaved = true; // device buffer will always be interleaved for nStreams > 1 and not mono mode\r
-- inChannels = stream_.nDeviceChannels[0];\r
-- }\r
--\r
-- if ( inInterleaved ) inOffset = 1;\r
-- else inOffset = stream_.bufferSize;\r
--\r
-- channelsLeft = inChannels;\r
-- for ( unsigned int i=0; i<handle->nStreams[0]; i++ ) {\r
-- in = inBuffer;\r
-- out = (Float32 *) outBufferList->mBuffers[handle->iStream[0]+i].mData;\r
-- streamChannels = outBufferList->mBuffers[handle->iStream[0]+i].mNumberChannels;\r
--\r
-- outJump = 0;\r
-- // Account for possible channel offset in first stream\r
-- if ( i == 0 && stream_.channelOffset[0] > 0 ) {\r
-- streamChannels -= stream_.channelOffset[0];\r
-- outJump = stream_.channelOffset[0];\r
-- out += outJump;\r
-- }\r
--\r
-- // Account for possible unfilled channels at end of the last stream\r
-- if ( streamChannels > channelsLeft ) {\r
-- outJump = streamChannels - channelsLeft;\r
-- streamChannels = channelsLeft;\r
-- }\r
--\r
-- // Determine input buffer offsets and skips\r
-- if ( inInterleaved ) {\r
-- inJump = inChannels;\r
-- in += inChannels - channelsLeft;\r
-- }\r
-- else {\r
-- inJump = 1;\r
-- in += (inChannels - channelsLeft) * inOffset;\r
-- }\r
--\r
-- for ( unsigned int i=0; i<stream_.bufferSize; i++ ) {\r
-- for ( unsigned int j=0; j<streamChannels; j++ ) {\r
-- *out++ = in[j*inOffset];\r
-- }\r
-- out += outJump;\r
-- in += inJump;\r
-- }\r
-- channelsLeft -= streamChannels;\r
-- }\r
-- }\r
-- }\r
-- }\r
--\r
-- // Don't bother draining input\r
-- if ( handle->drainCounter ) {\r
-- handle->drainCounter++;\r
-- goto unlock;\r
-- }\r
--\r
-- AudioDeviceID inputDevice;\r
-- inputDevice = handle->id[1];\r
-- if ( stream_.mode == INPUT || ( stream_.mode == DUPLEX && deviceId == inputDevice ) ) {\r
--\r
-- if ( handle->nStreams[1] == 1 ) {\r
-- if ( stream_.doConvertBuffer[1] ) { // convert directly from CoreAudio stream buffer\r
-- convertBuffer( stream_.userBuffer[1],\r
-- (char *) inBufferList->mBuffers[handle->iStream[1]].mData,\r
-- stream_.convertInfo[1] );\r
-- }\r
-- else { // copy to user buffer\r
-- memcpy( stream_.userBuffer[1],\r
-- inBufferList->mBuffers[handle->iStream[1]].mData,\r
-- inBufferList->mBuffers[handle->iStream[1]].mDataByteSize );\r
-- }\r
-- }\r
-- else { // read from multiple streams\r
-- Float32 *outBuffer = (Float32 *) stream_.userBuffer[1];\r
-- if ( stream_.doConvertBuffer[1] ) outBuffer = (Float32 *) stream_.deviceBuffer;\r
--\r
-- if ( stream_.deviceInterleaved[1] == false ) { // mono mode\r
-- UInt32 bufferBytes = inBufferList->mBuffers[handle->iStream[1]].mDataByteSize;\r
-- for ( unsigned int i=0; i<stream_.nUserChannels[1]; i++ ) {\r
-- memcpy( (void *)&outBuffer[i*stream_.bufferSize],\r
-- inBufferList->mBuffers[handle->iStream[1]+i].mData, bufferBytes );\r
-- }\r
-- }\r
-- else { // read from multiple multi-channel streams\r
-- UInt32 streamChannels, channelsLeft, inJump, outJump, outOffset;\r
-- Float32 *out, *in;\r
--\r
-- bool outInterleaved = ( stream_.userInterleaved ) ? true : false;\r
-- UInt32 outChannels = stream_.nUserChannels[1];\r
-- if ( stream_.doConvertBuffer[1] ) {\r
-- outInterleaved = true; // device buffer will always be interleaved for nStreams > 1 and not mono mode\r
-- outChannels = stream_.nDeviceChannels[1];\r
-- }\r
--\r
-- if ( outInterleaved ) outOffset = 1;\r
-- else outOffset = stream_.bufferSize;\r
--\r
-- channelsLeft = outChannels;\r
-- for ( unsigned int i=0; i<handle->nStreams[1]; i++ ) {\r
-- out = outBuffer;\r
-- in = (Float32 *) inBufferList->mBuffers[handle->iStream[1]+i].mData;\r
-- streamChannels = inBufferList->mBuffers[handle->iStream[1]+i].mNumberChannels;\r
--\r
-- inJump = 0;\r
-- // Account for possible channel offset in first stream\r
-- if ( i == 0 && stream_.channelOffset[1] > 0 ) {\r
-- streamChannels -= stream_.channelOffset[1];\r
-- inJump = stream_.channelOffset[1];\r
-- in += inJump;\r
-- }\r
--\r
-- // Account for possible unread channels at end of the last stream\r
-- if ( streamChannels > channelsLeft ) {\r
-- inJump = streamChannels - channelsLeft;\r
-- streamChannels = channelsLeft;\r
-- }\r
--\r
-- // Determine output buffer offsets and skips\r
-- if ( outInterleaved ) {\r
-- outJump = outChannels;\r
-- out += outChannels - channelsLeft;\r
-- }\r
-- else {\r
-- outJump = 1;\r
-- out += (outChannels - channelsLeft) * outOffset;\r
-- }\r
--\r
-- for ( unsigned int i=0; i<stream_.bufferSize; i++ ) {\r
-- for ( unsigned int j=0; j<streamChannels; j++ ) {\r
-- out[j*outOffset] = *in++;\r
-- }\r
-- out += outJump;\r
-- in += inJump;\r
-- }\r
-- channelsLeft -= streamChannels;\r
-- }\r
-- }\r
--\r
-- if ( stream_.doConvertBuffer[1] ) { // convert from our internal "device" buffer\r
-- convertBuffer( stream_.userBuffer[1],\r
-- stream_.deviceBuffer,\r
-- stream_.convertInfo[1] );\r
-- }\r
-- }\r
-- }\r
--\r
-- unlock:\r
-- //MUTEX_UNLOCK( &stream_.mutex );\r
--\r
-- RtApi::tickStreamTime();\r
-- return SUCCESS;\r
--}\r
--\r
--const char* RtApiCore :: getErrorCode( OSStatus code )\r
--{\r
-- switch( code ) {\r
--\r
-- case kAudioHardwareNotRunningError:\r
-- return "kAudioHardwareNotRunningError";\r
--\r
-- case kAudioHardwareUnspecifiedError:\r
-- return "kAudioHardwareUnspecifiedError";\r
--\r
-- case kAudioHardwareUnknownPropertyError:\r
-- return "kAudioHardwareUnknownPropertyError";\r
--\r
-- case kAudioHardwareBadPropertySizeError:\r
-- return "kAudioHardwareBadPropertySizeError";\r
--\r
-- case kAudioHardwareIllegalOperationError:\r
-- return "kAudioHardwareIllegalOperationError";\r
--\r
-- case kAudioHardwareBadObjectError:\r
-- return "kAudioHardwareBadObjectError";\r
--\r
-- case kAudioHardwareBadDeviceError:\r
-- return "kAudioHardwareBadDeviceError";\r
--\r
-- case kAudioHardwareBadStreamError:\r
-- return "kAudioHardwareBadStreamError";\r
--\r
-- case kAudioHardwareUnsupportedOperationError:\r
-- return "kAudioHardwareUnsupportedOperationError";\r
--\r
-- case kAudioDeviceUnsupportedFormatError:\r
-- return "kAudioDeviceUnsupportedFormatError";\r
--\r
-- case kAudioDevicePermissionsError:\r
-- return "kAudioDevicePermissionsError";\r
--\r
-- default:\r
-- return "CoreAudio unknown error";\r
-- }\r
--}\r
--\r
-- //******************** End of __MACOSX_CORE__ *********************//\r
--#endif\r
--\r
--#if defined(__UNIX_JACK__)\r
--\r
--// JACK is a low-latency audio server, originally written for the\r
--// GNU/Linux operating system and now also ported to OS-X. It can\r
--// connect a number of different applications to an audio device, as\r
--// well as allowing them to share audio between themselves.\r
--//\r
--// When using JACK with RtAudio, "devices" refer to JACK clients that\r
--// have ports connected to the server. The JACK server is typically\r
--// started in a terminal as follows:\r
--//\r
--// .jackd -d alsa -d hw:0\r
--//\r
--// or through an interface program such as qjackctl. Many of the\r
--// parameters normally set for a stream are fixed by the JACK server\r
--// and can be specified when the JACK server is started. In\r
--// particular,\r
--//\r
--// .jackd -d alsa -d hw:0 -r 44100 -p 512 -n 4\r
--//\r
--// specifies a sample rate of 44100 Hz, a buffer size of 512 sample\r
--// frames, and number of buffers = 4. Once the server is running, it\r
--// is not possible to override these values. If the values are not\r
--// specified in the command-line, the JACK server uses default values.\r
--//\r
--// The JACK server does not have to be running when an instance of\r
--// RtApiJack is created, though the function getDeviceCount() will\r
--// report 0 devices found until JACK has been started. When no\r
--// devices are available (i.e., the JACK server is not running), a\r
--// stream cannot be opened.\r
--\r
--#include <jack/jack.h>\r
--#include <unistd.h>\r
--#include <cstdio>\r
--\r
--// A structure to hold various information related to the Jack API\r
--// implementation.\r
--struct JackHandle {\r
-- jack_client_t *client;\r
-- jack_port_t **ports[2];\r
-- std::string deviceName[2];\r
-- bool xrun[2];\r
-- pthread_cond_t condition;\r
-- int drainCounter; // Tracks callback counts when draining\r
-- bool internalDrain; // Indicates if stop is initiated from callback or not.\r
--\r
-- JackHandle()\r
-- :client(0), drainCounter(0), internalDrain(false) { ports[0] = 0; ports[1] = 0; xrun[0] = false; xrun[1] = false; }\r
--};\r
--\r
--/* --- Monocasual hack ------------------------------------------------------ */\r
--#if defined(__linux__) || defined(__FreeBSD__)\r
--void *RtApi :: __HACK__getJackClient() {\r
-- JackHandle *handle = (JackHandle *) stream_.apiHandle;\r
-- return (void*) handle->client;\r
--}\r
--#endif\r
--/* -------------------------------------------------------------------------- */\r
--\r
--static void jackSilentError( const char * ) {}\r
--\r
--RtApiJack :: RtApiJack()\r
--{\r
-- // Nothing to do here.\r
--#if !defined(__RTAUDIO_DEBUG__)\r
-- // Turn off Jack's internal error reporting.\r
-- jack_set_error_function( &jackSilentError );\r
--#endif\r
--}\r
--\r
--RtApiJack :: ~RtApiJack()\r
--{\r
-- if ( stream_.state != STREAM_CLOSED ) closeStream();\r
--}\r
--\r
--unsigned int RtApiJack :: getDeviceCount( void )\r
--{\r
-- // See if we can become a jack client.\r
-- jack_options_t options = (jack_options_t) ( JackNoStartServer ); //JackNullOption;\r
-- jack_status_t *status = NULL;\r
-- jack_client_t *client = jack_client_open( "RtApiJackCount", options, status );\r
-- if ( client == 0 ) return 0;\r
--\r
-- const char **ports;\r
-- std::string port, previousPort;\r
-- unsigned int nChannels = 0, nDevices = 0;\r
-- ports = jack_get_ports( client, NULL, NULL, 0 );\r
-- if ( ports ) {\r
-- // Parse the port names up to the first colon (:).\r
-- size_t iColon = 0;\r
-- do {\r
-- port = (char *) ports[ nChannels ];\r
-- iColon = port.find(":");\r
-- if ( iColon != std::string::npos ) {\r
-- port = port.substr( 0, iColon + 1 );\r
-- if ( port != previousPort ) {\r
-- nDevices++;\r
-- previousPort = port;\r
-- }\r
-- }\r
-- } while ( ports[++nChannels] );\r
-- free( ports );\r
-- }\r
--\r
-- jack_client_close( client );\r
-- return nDevices;\r
--}\r
--\r
--RtAudio::DeviceInfo RtApiJack :: getDeviceInfo( unsigned int device )\r
--{\r
-- RtAudio::DeviceInfo info;\r
-- info.probed = false;\r
--\r
-- jack_options_t options = (jack_options_t) ( JackNoStartServer ); //JackNullOption\r
-- jack_status_t *status = NULL;\r
-- jack_client_t *client = jack_client_open( "RtApiJackInfo", options, status );\r
-- if ( client == 0 ) {\r
-- errorText_ = "RtApiJack::getDeviceInfo: Jack server not found or connection error!";\r
-- error( RtAudioError::WARNING );\r
-- return info;\r
-- }\r
--\r
-- const char **ports;\r
-- std::string port, previousPort;\r
-- unsigned int nPorts = 0, nDevices = 0;\r
-- ports = jack_get_ports( client, NULL, NULL, 0 );\r
-- if ( ports ) {\r
-- // Parse the port names up to the first colon (:).\r
-- size_t iColon = 0;\r
-- do {\r
-- port = (char *) ports[ nPorts ];\r
-- iColon = port.find(":");\r
-- if ( iColon != std::string::npos ) {\r
-- port = port.substr( 0, iColon );\r
-- if ( port != previousPort ) {\r
-- if ( nDevices == device ) info.name = port;\r
-- nDevices++;\r
-- previousPort = port;\r
-- }\r
-- }\r
-- } while ( ports[++nPorts] );\r
-- free( ports );\r
-- }\r
--\r
-- if ( device >= nDevices ) {\r
-- jack_client_close( client );\r
-- errorText_ = "RtApiJack::getDeviceInfo: device ID is invalid!";\r
-- error( RtAudioError::INVALID_USE );\r
-- return info;\r
-- }\r
--\r
-- // Get the current jack server sample rate.\r
-- info.sampleRates.clear();\r
--\r
-- info.preferredSampleRate = jack_get_sample_rate( client );\r
-- info.sampleRates.push_back( info.preferredSampleRate );\r
--\r
-- // Count the available ports containing the client name as device\r
-- // channels. Jack "input ports" equal RtAudio output channels.\r
-- unsigned int nChannels = 0;\r
-- ports = jack_get_ports( client, info.name.c_str(), NULL, JackPortIsInput );\r
-- if ( ports ) {\r
-- while ( ports[ nChannels ] ) nChannels++;\r
-- free( ports );\r
-- info.outputChannels = nChannels;\r
-- }\r
--\r
-- // Jack "output ports" equal RtAudio input channels.\r
-- nChannels = 0;\r
-- ports = jack_get_ports( client, info.name.c_str(), NULL, JackPortIsOutput );\r
-- if ( ports ) {\r
-- while ( ports[ nChannels ] ) nChannels++;\r
-- free( ports );\r
-- info.inputChannels = nChannels;\r
-- }\r
--\r
-- if ( info.outputChannels == 0 && info.inputChannels == 0 ) {\r
-- jack_client_close(client);\r
-- errorText_ = "RtApiJack::getDeviceInfo: error determining Jack input/output channels!";\r
-- error( RtAudioError::WARNING );\r
-- return info;\r
-- }\r
--\r
-- // If device opens for both playback and capture, we determine the channels.\r
-- if ( info.outputChannels > 0 && info.inputChannels > 0 )\r
-- info.duplexChannels = (info.outputChannels > info.inputChannels) ? info.inputChannels : info.outputChannels;\r
--\r
-- // Jack always uses 32-bit floats.\r
-- info.nativeFormats = RTAUDIO_FLOAT32;\r
--\r
-- // Jack doesn't provide default devices so we'll use the first available one.\r
-- if ( device == 0 && info.outputChannels > 0 )\r
-- info.isDefaultOutput = true;\r
-- if ( device == 0 && info.inputChannels > 0 )\r
-- info.isDefaultInput = true;\r
--\r
-- jack_client_close(client);\r
-- info.probed = true;\r
-- return info;\r
--}\r
--\r
--static int jackCallbackHandler( jack_nframes_t nframes, void *infoPointer )\r
--{\r
-- CallbackInfo *info = (CallbackInfo *) infoPointer;\r
--\r
-- RtApiJack *object = (RtApiJack *) info->object;\r
-- if ( object->callbackEvent( (unsigned long) nframes ) == false ) return 1;\r
--\r
-- return 0;\r
--}\r
--\r
--// This function will be called by a spawned thread when the Jack\r
--// server signals that it is shutting down. It is necessary to handle\r
--// it this way because the jackShutdown() function must return before\r
--// the jack_deactivate() function (in closeStream()) will return.\r
--static void *jackCloseStream( void *ptr )\r
--{\r
-- CallbackInfo *info = (CallbackInfo *) ptr;\r
-- RtApiJack *object = (RtApiJack *) info->object;\r
--\r
-- object->closeStream();\r
--\r
-- pthread_exit( NULL );\r
--}\r
--static void jackShutdown( void *infoPointer )\r
--{\r
-- CallbackInfo *info = (CallbackInfo *) infoPointer;\r
-- RtApiJack *object = (RtApiJack *) info->object;\r
--\r
-- // Check current stream state. If stopped, then we'll assume this\r
-- // was called as a result of a call to RtApiJack::stopStream (the\r
-- // deactivation of a client handle causes this function to be called).\r
-- // If not, we'll assume the Jack server is shutting down or some\r
-- // other problem occurred and we should close the stream.\r
-- if ( object->isStreamRunning() == false ) return;\r
--\r
-- ThreadHandle threadId;\r
-- pthread_create( &threadId, NULL, jackCloseStream, info );\r
-- std::cerr << "\nRtApiJack: the Jack server is shutting down this client ... stream stopped and closed!!\n" << std::endl;\r
--}\r
--\r
--static int jackXrun( void *infoPointer )\r
--{\r
-- JackHandle *handle = (JackHandle *) infoPointer;\r
--\r
-- if ( handle->ports[0] ) handle->xrun[0] = true;\r
-- if ( handle->ports[1] ) handle->xrun[1] = true;\r
--\r
-- return 0;\r
--}\r
--\r
--bool RtApiJack :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,\r
-- unsigned int firstChannel, unsigned int sampleRate,\r
-- RtAudioFormat format, unsigned int *bufferSize,\r
-- RtAudio::StreamOptions *options )\r
--{\r
-- JackHandle *handle = (JackHandle *) stream_.apiHandle;\r
--\r
-- // Look for jack server and try to become a client (only do once per stream).\r
-- jack_client_t *client = 0;\r
-- if ( mode == OUTPUT || ( mode == INPUT && stream_.mode != OUTPUT ) ) {\r
-- jack_options_t jackoptions = (jack_options_t) ( JackNoStartServer ); //JackNullOption;\r
-- jack_status_t *status = NULL;\r
-- if ( options && !options->streamName.empty() )\r
-- client = jack_client_open( options->streamName.c_str(), jackoptions, status );\r
-- else\r
-- client = jack_client_open( "RtApiJack", jackoptions, status );\r
-- if ( client == 0 ) {\r
-- errorText_ = "RtApiJack::probeDeviceOpen: Jack server not found or connection error!";\r
-- error( RtAudioError::WARNING );\r
-- return FAILURE;\r
-- }\r
-- }\r
-- else {\r
-- // The handle must have been created on an earlier pass.\r
-- client = handle->client;\r
-- }\r
--\r
-- const char **ports;\r
-- std::string port, previousPort, deviceName;\r
-- unsigned int nPorts = 0, nDevices = 0;\r
-- ports = jack_get_ports( client, NULL, NULL, 0 );\r
-- if ( ports ) {\r
-- // Parse the port names up to the first colon (:).\r
-- size_t iColon = 0;\r
-- do {\r
-- port = (char *) ports[ nPorts ];\r
-- iColon = port.find(":");\r
-- if ( iColon != std::string::npos ) {\r
-- port = port.substr( 0, iColon );\r
-- if ( port != previousPort ) {\r
-- if ( nDevices == device ) deviceName = port;\r
-- nDevices++;\r
-- previousPort = port;\r
-- }\r
-- }\r
-- } while ( ports[++nPorts] );\r
-- free( ports );\r
-- }\r
--\r
-- if ( device >= nDevices ) {\r
-- errorText_ = "RtApiJack::probeDeviceOpen: device ID is invalid!";\r
-- return FAILURE;\r
-- }\r
--\r
-- // Count the available ports containing the client name as device\r
-- // channels. Jack "input ports" equal RtAudio output channels.\r
-- unsigned int nChannels = 0;\r
-- unsigned long flag = JackPortIsInput;\r
-- if ( mode == INPUT ) flag = JackPortIsOutput;\r
-- ports = jack_get_ports( client, deviceName.c_str(), NULL, flag );\r
-- if ( ports ) {\r
-- while ( ports[ nChannels ] ) nChannels++;\r
-- free( ports );\r
-- }\r
--\r
-- // Compare the jack ports for specified client to the requested number of channels.\r
-- if ( nChannels < (channels + firstChannel) ) {\r
-- errorStream_ << "RtApiJack::probeDeviceOpen: requested number of channels (" << channels << ") + offset (" << firstChannel << ") not found for specified device (" << device << ":" << deviceName << ").";\r
-- errorText_ = errorStream_.str();\r
-- return FAILURE;\r
-- }\r
--\r
-- // Check the jack server sample rate.\r
-- unsigned int jackRate = jack_get_sample_rate( client );\r
-- if ( sampleRate != jackRate ) {\r
-- jack_client_close( client );\r
-- errorStream_ << "RtApiJack::probeDeviceOpen: the requested sample rate (" << sampleRate << ") is different than the JACK server rate (" << jackRate << ").";\r
-- errorText_ = errorStream_.str();\r
-- return FAILURE;\r
-- }\r
-- stream_.sampleRate = jackRate;\r
--\r
-- // Get the latency of the JACK port.\r
-- ports = jack_get_ports( client, deviceName.c_str(), NULL, flag );\r
-- if ( ports[ firstChannel ] ) {\r
-- // Added by Ge Wang\r
-- jack_latency_callback_mode_t cbmode = (mode == INPUT ? JackCaptureLatency : JackPlaybackLatency);\r
-- // the range (usually the min and max are equal)\r
-- jack_latency_range_t latrange; latrange.min = latrange.max = 0;\r
-- // get the latency range\r
-- jack_port_get_latency_range( jack_port_by_name( client, ports[firstChannel] ), cbmode, &latrange );\r
-- // be optimistic, use the min!\r
-- stream_.latency[mode] = latrange.min;\r
-- //stream_.latency[mode] = jack_port_get_latency( jack_port_by_name( client, ports[ firstChannel ] ) );\r
-- }\r
-- free( ports );\r
--\r
-- // The jack server always uses 32-bit floating-point data.\r
-- stream_.deviceFormat[mode] = RTAUDIO_FLOAT32;\r
-- stream_.userFormat = format;\r
--\r
-- if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) stream_.userInterleaved = false;\r
-- else stream_.userInterleaved = true;\r
--\r
-- // Jack always uses non-interleaved buffers.\r
-- stream_.deviceInterleaved[mode] = false;\r
--\r
-- // Jack always provides host byte-ordered data.\r
-- stream_.doByteSwap[mode] = false;\r
--\r
-- // Get the buffer size. The buffer size and number of buffers\r
-- // (periods) is set when the jack server is started.\r
-- stream_.bufferSize = (int) jack_get_buffer_size( client );\r
-- *bufferSize = stream_.bufferSize;\r
--\r
-- stream_.nDeviceChannels[mode] = channels;\r
-- stream_.nUserChannels[mode] = channels;\r
--\r
-- // Set flags for buffer conversion.\r
-- stream_.doConvertBuffer[mode] = false;\r
-- if ( stream_.userFormat != stream_.deviceFormat[mode] )\r
-- stream_.doConvertBuffer[mode] = true;\r
-- if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] &&\r
-- stream_.nUserChannels[mode] > 1 )\r
-- stream_.doConvertBuffer[mode] = true;\r
--\r
-- // Allocate our JackHandle structure for the stream.\r
-- if ( handle == 0 ) {\r
-- try {\r
-- handle = new JackHandle;\r
-- }\r
-- catch ( std::bad_alloc& ) {\r
-- errorText_ = "RtApiJack::probeDeviceOpen: error allocating JackHandle memory.";\r
-- goto error;\r
-- }\r
--\r
-- if ( pthread_cond_init(&handle->condition, NULL) ) {\r
-- errorText_ = "RtApiJack::probeDeviceOpen: error initializing pthread condition variable.";\r
-- goto error;\r
-- }\r
-- stream_.apiHandle = (void *) handle;\r
-- handle->client = client;\r
-- }\r
-- handle->deviceName[mode] = deviceName;\r
--\r
-- // Allocate necessary internal buffers.\r
-- unsigned long bufferBytes;\r
-- bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat );\r
-- stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 );\r
-- if ( stream_.userBuffer[mode] == NULL ) {\r
-- errorText_ = "RtApiJack::probeDeviceOpen: error allocating user buffer memory.";\r
-- goto error;\r
-- }\r
--\r
-- if ( stream_.doConvertBuffer[mode] ) {\r
--\r
-- bool makeBuffer = true;\r
-- if ( mode == OUTPUT )\r
-- bufferBytes = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] );\r
-- else { // mode == INPUT\r
-- bufferBytes = stream_.nDeviceChannels[1] * formatBytes( stream_.deviceFormat[1] );\r
-- if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) {\r
-- unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes(stream_.deviceFormat[0]);\r
-- if ( bufferBytes < bytesOut ) makeBuffer = false;\r
-- }\r
-- }\r
--\r
-- if ( makeBuffer ) {\r
-- bufferBytes *= *bufferSize;\r
-- if ( stream_.deviceBuffer ) free( stream_.deviceBuffer );\r
-- stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 );\r
-- if ( stream_.deviceBuffer == NULL ) {\r
-- errorText_ = "RtApiJack::probeDeviceOpen: error allocating device buffer memory.";\r
-- goto error;\r
-- }\r
-- }\r
-- }\r
--\r
-- // Allocate memory for the Jack ports (channels) identifiers.\r
-- handle->ports[mode] = (jack_port_t **) malloc ( sizeof (jack_port_t *) * channels );\r
-- if ( handle->ports[mode] == NULL ) {\r
-- errorText_ = "RtApiJack::probeDeviceOpen: error allocating port memory.";\r
-- goto error;\r
-- }\r
--\r
-- stream_.device[mode] = device;\r
-- stream_.channelOffset[mode] = firstChannel;\r
-- stream_.state = STREAM_STOPPED;\r
-- stream_.callbackInfo.object = (void *) this;\r
--\r
-- if ( stream_.mode == OUTPUT && mode == INPUT )\r
-- // We had already set up the stream for output.\r
-- stream_.mode = DUPLEX;\r
-- else {\r
-- stream_.mode = mode;\r
-- jack_set_process_callback( handle->client, jackCallbackHandler, (void *) &stream_.callbackInfo );\r
-- jack_set_xrun_callback( handle->client, jackXrun, (void *) &handle );\r
-- jack_on_shutdown( handle->client, jackShutdown, (void *) &stream_.callbackInfo );\r
-- }\r
--\r
-- // Register our ports.\r
-- char label[64];\r
-- if ( mode == OUTPUT ) {\r
-- for ( unsigned int i=0; i<stream_.nUserChannels[0]; i++ ) {\r
-- snprintf( label, 64, "outport %d", i );\r
-- handle->ports[0][i] = jack_port_register( handle->client, (const char *)label,\r
-- JACK_DEFAULT_AUDIO_TYPE, JackPortIsOutput, 0 );\r
-- }\r
-- }\r
-- else {\r
-- for ( unsigned int i=0; i<stream_.nUserChannels[1]; i++ ) {\r
-- snprintf( label, 64, "inport %d", i );\r
-- handle->ports[1][i] = jack_port_register( handle->client, (const char *)label,\r
-- JACK_DEFAULT_AUDIO_TYPE, JackPortIsInput, 0 );\r
-- }\r
-- }\r
--\r
-- // Setup the buffer conversion information structure. We don't use\r
-- // buffers to do channel offsets, so we override that parameter\r
-- // here.\r
-- if ( stream_.doConvertBuffer[mode] ) setConvertInfo( mode, 0 );\r
--\r
-- return SUCCESS;\r
--\r
-- error:\r
-- if ( handle ) {\r
-- pthread_cond_destroy( &handle->condition );\r
-- jack_client_close( handle->client );\r
--\r
-- if ( handle->ports[0] ) free( handle->ports[0] );\r
-- if ( handle->ports[1] ) free( handle->ports[1] );\r
--\r
-- delete handle;\r
-- stream_.apiHandle = 0;\r
-- }\r
--\r
-- for ( int i=0; i<2; i++ ) {\r
-- if ( stream_.userBuffer[i] ) {\r
-- free( stream_.userBuffer[i] );\r
-- stream_.userBuffer[i] = 0;\r
-- }\r
-- }\r
--\r
-- if ( stream_.deviceBuffer ) {\r
-- free( stream_.deviceBuffer );\r
-- stream_.deviceBuffer = 0;\r
-- }\r
--\r
-- return FAILURE;\r
--}\r
--\r
--void RtApiJack :: closeStream( void )\r
--{\r
-- if ( stream_.state == STREAM_CLOSED ) {\r
-- errorText_ = "RtApiJack::closeStream(): no open stream to close!";\r
-- error( RtAudioError::WARNING );\r
-- return;\r
-- }\r
--\r
-- JackHandle *handle = (JackHandle *) stream_.apiHandle;\r
-- if ( handle ) {\r
--\r
-- if ( stream_.state == STREAM_RUNNING )\r
-- jack_deactivate( handle->client );\r
--\r
-- jack_client_close( handle->client );\r
-- }\r
--\r
-- if ( handle ) {\r
-- if ( handle->ports[0] ) free( handle->ports[0] );\r
-- if ( handle->ports[1] ) free( handle->ports[1] );\r
-- pthread_cond_destroy( &handle->condition );\r
-- delete handle;\r
-- stream_.apiHandle = 0;\r
-- }\r
--\r
-- for ( int i=0; i<2; i++ ) {\r
-- if ( stream_.userBuffer[i] ) {\r
-- free( stream_.userBuffer[i] );\r
-- stream_.userBuffer[i] = 0;\r
-- }\r
-- }\r
--\r
-- if ( stream_.deviceBuffer ) {\r
-- free( stream_.deviceBuffer );\r
-- stream_.deviceBuffer = 0;\r
-- }\r
--\r
-- stream_.mode = UNINITIALIZED;\r
-- stream_.state = STREAM_CLOSED;\r
--}\r
--\r
--void RtApiJack :: startStream( void )\r
--{\r
-- verifyStream();\r
-- if ( stream_.state == STREAM_RUNNING ) {\r
-- errorText_ = "RtApiJack::startStream(): the stream is already running!";\r
-- error( RtAudioError::WARNING );\r
-- return;\r
-- }\r
--\r
-- JackHandle *handle = (JackHandle *) stream_.apiHandle;\r
-- int result = jack_activate( handle->client );\r
-- if ( result ) {\r
-- errorText_ = "RtApiJack::startStream(): unable to activate JACK client!";\r
-- goto unlock;\r
-- }\r
--\r
-- const char **ports;\r
--\r
-- // Get the list of available ports.\r
-- if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {\r
-- result = 1;\r
-- ports = jack_get_ports( handle->client, handle->deviceName[0].c_str(), NULL, JackPortIsInput);\r
-- if ( ports == NULL) {\r
-- errorText_ = "RtApiJack::startStream(): error determining available JACK input ports!";\r
-- goto unlock;\r
-- }\r
--\r
-- // Now make the port connections. Since RtAudio wasn't designed to\r
-- // allow the user to select particular channels of a device, we'll\r
-- // just open the first "nChannels" ports with offset.\r
-- for ( unsigned int i=0; i<stream_.nUserChannels[0]; i++ ) {\r
-- result = 1;\r
-- if ( ports[ stream_.channelOffset[0] + i ] )\r
-- result = jack_connect( handle->client, jack_port_name( handle->ports[0][i] ), ports[ stream_.channelOffset[0] + i ] );\r
-- if ( result ) {\r
-- free( ports );\r
-- errorText_ = "RtApiJack::startStream(): error connecting output ports!";\r
-- goto unlock;\r
-- }\r
-- }\r
-- free(ports);\r
-- }\r
--\r
-- if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {\r
-- result = 1;\r
-- ports = jack_get_ports( handle->client, handle->deviceName[1].c_str(), NULL, JackPortIsOutput );\r
-- if ( ports == NULL) {\r
-- errorText_ = "RtApiJack::startStream(): error determining available JACK output ports!";\r
-- goto unlock;\r
-- }\r
--\r
-- // Now make the port connections. See note above.\r
-- for ( unsigned int i=0; i<stream_.nUserChannels[1]; i++ ) {\r
-- result = 1;\r
-- if ( ports[ stream_.channelOffset[1] + i ] )\r
-- result = jack_connect( handle->client, ports[ stream_.channelOffset[1] + i ], jack_port_name( handle->ports[1][i] ) );\r
-- if ( result ) {\r
-- free( ports );\r
-- errorText_ = "RtApiJack::startStream(): error connecting input ports!";\r
-- goto unlock;\r
-- }\r
-- }\r
-- free(ports);\r
-- }\r
--\r
-- handle->drainCounter = 0;\r
-- handle->internalDrain = false;\r
-- stream_.state = STREAM_RUNNING;\r
--\r
-- unlock:\r
-- if ( result == 0 ) return;\r
-- error( RtAudioError::SYSTEM_ERROR );\r
--}\r
--\r
--void RtApiJack :: stopStream( void )\r
--{\r
-- verifyStream();\r
-- if ( stream_.state == STREAM_STOPPED ) {\r
-- errorText_ = "RtApiJack::stopStream(): the stream is already stopped!";\r
-- error( RtAudioError::WARNING );\r
-- return;\r
-- }\r
--\r
-- JackHandle *handle = (JackHandle *) stream_.apiHandle;\r
-- if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {\r
--\r
-- if ( handle->drainCounter == 0 ) {\r
-- handle->drainCounter = 2;\r
-- pthread_cond_wait( &handle->condition, &stream_.mutex ); // block until signaled\r
-- }\r
-- }\r
--\r
-- jack_deactivate( handle->client );\r
-- stream_.state = STREAM_STOPPED;\r
--}\r
--\r
--void RtApiJack :: abortStream( void )\r
--{\r
-- verifyStream();\r
-- if ( stream_.state == STREAM_STOPPED ) {\r
-- errorText_ = "RtApiJack::abortStream(): the stream is already stopped!";\r
-- error( RtAudioError::WARNING );\r
-- return;\r
-- }\r
--\r
-- JackHandle *handle = (JackHandle *) stream_.apiHandle;\r
-- handle->drainCounter = 2;\r
--\r
-- stopStream();\r
--}\r
--\r
--// This function will be called by a spawned thread when the user\r
--// callback function signals that the stream should be stopped or\r
--// aborted. It is necessary to handle it this way because the\r
--// callbackEvent() function must return before the jack_deactivate()\r
--// function will return.\r
--static void *jackStopStream( void *ptr )\r
--{\r
-- CallbackInfo *info = (CallbackInfo *) ptr;\r
-- RtApiJack *object = (RtApiJack *) info->object;\r
--\r
-- object->stopStream();\r
-- pthread_exit( NULL );\r
--}\r
--\r
--bool RtApiJack :: callbackEvent( unsigned long nframes )\r
--{\r
-- if ( stream_.state == STREAM_STOPPED || stream_.state == STREAM_STOPPING ) return SUCCESS;\r
-- if ( stream_.state == STREAM_CLOSED ) {\r
-- errorText_ = "RtApiCore::callbackEvent(): the stream is closed ... this shouldn't happen!";\r
-- error( RtAudioError::WARNING );\r
-- return FAILURE;\r
-- }\r
-- if ( stream_.bufferSize != nframes ) {\r
-- errorText_ = "RtApiCore::callbackEvent(): the JACK buffer size has changed ... cannot process!";\r
-- error( RtAudioError::WARNING );\r
-- return FAILURE;\r
-- }\r
--\r
-- CallbackInfo *info = (CallbackInfo *) &stream_.callbackInfo;\r
-- JackHandle *handle = (JackHandle *) stream_.apiHandle;\r
--\r
-- // Check if we were draining the stream and signal is finished.\r
-- if ( handle->drainCounter > 3 ) {\r
-- ThreadHandle threadId;\r
--\r
-- stream_.state = STREAM_STOPPING;\r
-- if ( handle->internalDrain == true )\r
-- pthread_create( &threadId, NULL, jackStopStream, info );\r
-- else\r
-- pthread_cond_signal( &handle->condition );\r
-- return SUCCESS;\r
-- }\r
--\r
-- // Invoke user callback first, to get fresh output data.\r
-- if ( handle->drainCounter == 0 ) {\r
-- RtAudioCallback callback = (RtAudioCallback) info->callback;\r
-- double streamTime = getStreamTime();\r
-- RtAudioStreamStatus status = 0;\r
-- if ( stream_.mode != INPUT && handle->xrun[0] == true ) {\r
-- status |= RTAUDIO_OUTPUT_UNDERFLOW;\r
-- handle->xrun[0] = false;\r
-- }\r
-- if ( stream_.mode != OUTPUT && handle->xrun[1] == true ) {\r
-- status |= RTAUDIO_INPUT_OVERFLOW;\r
-- handle->xrun[1] = false;\r
-- }\r
-- int cbReturnValue = callback( stream_.userBuffer[0], stream_.userBuffer[1],\r
-- stream_.bufferSize, streamTime, status, info->userData );\r
-- if ( cbReturnValue == 2 ) {\r
-- stream_.state = STREAM_STOPPING;\r
-- handle->drainCounter = 2;\r
-- ThreadHandle id;\r
-- pthread_create( &id, NULL, jackStopStream, info );\r
-- return SUCCESS;\r
-- }\r
-- else if ( cbReturnValue == 1 ) {\r
-- handle->drainCounter = 1;\r
-- handle->internalDrain = true;\r
-- }\r
-- }\r
--\r
-- jack_default_audio_sample_t *jackbuffer;\r
-- unsigned long bufferBytes = nframes * sizeof( jack_default_audio_sample_t );\r
-- if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {\r
--\r
-- if ( handle->drainCounter > 1 ) { // write zeros to the output stream\r
--\r
-- for ( unsigned int i=0; i<stream_.nDeviceChannels[0]; i++ ) {\r
-- jackbuffer = (jack_default_audio_sample_t *) jack_port_get_buffer( handle->ports[0][i], (jack_nframes_t) nframes );\r
-- memset( jackbuffer, 0, bufferBytes );\r
-- }\r
--\r
-- }\r
-- else if ( stream_.doConvertBuffer[0] ) {\r
--\r
-- convertBuffer( stream_.deviceBuffer, stream_.userBuffer[0], stream_.convertInfo[0] );\r
--\r
-- for ( unsigned int i=0; i<stream_.nDeviceChannels[0]; i++ ) {\r
-- jackbuffer = (jack_default_audio_sample_t *) jack_port_get_buffer( handle->ports[0][i], (jack_nframes_t) nframes );\r
-- memcpy( jackbuffer, &stream_.deviceBuffer[i*bufferBytes], bufferBytes );\r
-- }\r
-- }\r
-- else { // no buffer conversion\r
-- for ( unsigned int i=0; i<stream_.nUserChannels[0]; i++ ) {\r
-- jackbuffer = (jack_default_audio_sample_t *) jack_port_get_buffer( handle->ports[0][i], (jack_nframes_t) nframes );\r
-- memcpy( jackbuffer, &stream_.userBuffer[0][i*bufferBytes], bufferBytes );\r
-- }\r
-- }\r
-- }\r
--\r
-- // Don't bother draining input\r
-- if ( handle->drainCounter ) {\r
-- handle->drainCounter++;\r
-- goto unlock;\r
-- }\r
--\r
-- if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {\r
--\r
-- if ( stream_.doConvertBuffer[1] ) {\r
-- for ( unsigned int i=0; i<stream_.nDeviceChannels[1]; i++ ) {\r
-- jackbuffer = (jack_default_audio_sample_t *) jack_port_get_buffer( handle->ports[1][i], (jack_nframes_t) nframes );\r
-- memcpy( &stream_.deviceBuffer[i*bufferBytes], jackbuffer, bufferBytes );\r
-- }\r
-- convertBuffer( stream_.userBuffer[1], stream_.deviceBuffer, stream_.convertInfo[1] );\r
-- }\r
-- else { // no buffer conversion\r
-- for ( unsigned int i=0; i<stream_.nUserChannels[1]; i++ ) {\r
-- jackbuffer = (jack_default_audio_sample_t *) jack_port_get_buffer( handle->ports[1][i], (jack_nframes_t) nframes );\r
-- memcpy( &stream_.userBuffer[1][i*bufferBytes], jackbuffer, bufferBytes );\r
-- }\r
-- }\r
-- }\r
--\r
-- unlock:\r
-- RtApi::tickStreamTime();\r
-- return SUCCESS;\r
--}\r
-- //******************** End of __UNIX_JACK__ *********************//\r
--#endif\r
--\r
--#if defined(__WINDOWS_ASIO__) // ASIO API on Windows\r
--\r
--// The ASIO API is designed around a callback scheme, so this\r
--// implementation is similar to that used for OS-X CoreAudio and Linux\r
--// Jack. The primary constraint with ASIO is that it only allows\r
--// access to a single driver at a time. Thus, it is not possible to\r
--// have more than one simultaneous RtAudio stream.\r
--//\r
--// This implementation also requires a number of external ASIO files\r
--// and a few global variables. The ASIO callback scheme does not\r
--// allow for the passing of user data, so we must create a global\r
--// pointer to our callbackInfo structure.\r
--//\r
--// On unix systems, we make use of a pthread condition variable.\r
--// Since there is no equivalent in Windows, I hacked something based\r
--// on information found in\r
--// http://www.cs.wustl.edu/~schmidt/win32-cv-1.html.\r
--\r
--#include "asiosys.h"\r
--#include "asio.h"\r
--#include "iasiothiscallresolver.h"\r
--#include "asiodrivers.h"\r
--#include <cmath>\r
--\r
--static AsioDrivers drivers;\r
--static ASIOCallbacks asioCallbacks;\r
--static ASIODriverInfo driverInfo;\r
--static CallbackInfo *asioCallbackInfo;\r
--static bool asioXRun;\r
--\r
--struct AsioHandle {\r
-- int drainCounter; // Tracks callback counts when draining\r
-- bool internalDrain; // Indicates if stop is initiated from callback or not.\r
-- ASIOBufferInfo *bufferInfos;\r
-- HANDLE condition;\r
--\r
-- AsioHandle()\r
-- :drainCounter(0), internalDrain(false), bufferInfos(0) {}\r
--};\r
--\r
--// Function declarations (definitions at end of section)\r
--static const char* getAsioErrorString( ASIOError result );\r
--static void sampleRateChanged( ASIOSampleRate sRate );\r
--static long asioMessages( long selector, long value, void* message, double* opt );\r
--\r
--RtApiAsio :: RtApiAsio()\r
--{\r
-- // ASIO cannot run on a multi-threaded appartment. You can call\r
-- // CoInitialize beforehand, but it must be for appartment threading\r
-- // (in which case, CoInitilialize will return S_FALSE here).\r
-- coInitialized_ = false;\r
-- HRESULT hr = CoInitialize( NULL );\r
-- if ( FAILED(hr) ) {\r
-- errorText_ = "RtApiAsio::ASIO requires a single-threaded appartment. Call CoInitializeEx(0,COINIT_APARTMENTTHREADED)";\r
-- error( RtAudioError::WARNING );\r
-- }\r
-- coInitialized_ = true;\r
--\r
-- drivers.removeCurrentDriver();\r
-- driverInfo.asioVersion = 2;\r
--\r
-- // See note in DirectSound implementation about GetDesktopWindow().\r
-- driverInfo.sysRef = GetForegroundWindow();\r
--}\r
--\r
--RtApiAsio :: ~RtApiAsio()\r
--{\r
-- if ( stream_.state != STREAM_CLOSED ) closeStream();\r
-- if ( coInitialized_ ) CoUninitialize();\r
--}\r
--\r
--unsigned int RtApiAsio :: getDeviceCount( void )\r
--{\r
-- return (unsigned int) drivers.asioGetNumDev();\r
--}\r
--\r
--RtAudio::DeviceInfo RtApiAsio :: getDeviceInfo( unsigned int device )\r
--{\r
-- RtAudio::DeviceInfo info;\r
-- info.probed = false;\r
--\r
-- // Get device ID\r
-- unsigned int nDevices = getDeviceCount();\r
-- if ( nDevices == 0 ) {\r
-- errorText_ = "RtApiAsio::getDeviceInfo: no devices found!";\r
-- error( RtAudioError::INVALID_USE );\r
-- return info;\r
-- }\r
--\r
-- if ( device >= nDevices ) {\r
-- errorText_ = "RtApiAsio::getDeviceInfo: device ID is invalid!";\r
-- error( RtAudioError::INVALID_USE );\r
-- return info;\r
-- }\r
--\r
-- // If a stream is already open, we cannot probe other devices. Thus, use the saved results.\r
-- if ( stream_.state != STREAM_CLOSED ) {\r
-- if ( device >= devices_.size() ) {\r
-- errorText_ = "RtApiAsio::getDeviceInfo: device ID was not present before stream was opened.";\r
-- error( RtAudioError::WARNING );\r
-- return info;\r
-- }\r
-- return devices_[ device ];\r
-- }\r
--\r
-- char driverName[32];\r
-- ASIOError result = drivers.asioGetDriverName( (int) device, driverName, 32 );\r
-- if ( result != ASE_OK ) {\r
-- errorStream_ << "RtApiAsio::getDeviceInfo: unable to get driver name (" << getAsioErrorString( result ) << ").";\r
-- errorText_ = errorStream_.str();\r
-- error( RtAudioError::WARNING );\r
-- return info;\r
-- }\r
--\r
-- info.name = driverName;\r
--\r
-- if ( !drivers.loadDriver( driverName ) ) {\r
-- errorStream_ << "RtApiAsio::getDeviceInfo: unable to load driver (" << driverName << ").";\r
-- errorText_ = errorStream_.str();\r
-- error( RtAudioError::WARNING );\r
-- return info;\r
-- }\r
--\r
-- result = ASIOInit( &driverInfo );\r
-- if ( result != ASE_OK ) {\r
-- errorStream_ << "RtApiAsio::getDeviceInfo: error (" << getAsioErrorString( result ) << ") initializing driver (" << driverName << ").";\r
-- errorText_ = errorStream_.str();\r
-- error( RtAudioError::WARNING );\r
-- return info;\r
-- }\r
--\r
-- // Determine the device channel information.\r
-- long inputChannels, outputChannels;\r
-- result = ASIOGetChannels( &inputChannels, &outputChannels );\r
-- if ( result != ASE_OK ) {\r
-- drivers.removeCurrentDriver();\r
-- errorStream_ << "RtApiAsio::getDeviceInfo: error (" << getAsioErrorString( result ) << ") getting channel count (" << driverName << ").";\r
-- errorText_ = errorStream_.str();\r
-- error( RtAudioError::WARNING );\r
-- return info;\r
-- }\r
--\r
-- info.outputChannels = outputChannels;\r
-- info.inputChannels = inputChannels;\r
-- if ( info.outputChannels > 0 && info.inputChannels > 0 )\r
-- info.duplexChannels = (info.outputChannels > info.inputChannels) ? info.inputChannels : info.outputChannels;\r
--\r
-- // Determine the supported sample rates.\r
-- info.sampleRates.clear();\r
-- for ( unsigned int i=0; i<MAX_SAMPLE_RATES; i++ ) {\r
-- result = ASIOCanSampleRate( (ASIOSampleRate) SAMPLE_RATES[i] );\r
-- if ( result == ASE_OK ) {\r
-- info.sampleRates.push_back( SAMPLE_RATES[i] );\r
--\r
-- if ( !info.preferredSampleRate || ( SAMPLE_RATES[i] <= 48000 && SAMPLE_RATES[i] > info.preferredSampleRate ) )\r
-- info.preferredSampleRate = SAMPLE_RATES[i];\r
-- }\r
-- }\r
--\r
-- // Determine supported data types ... just check first channel and assume rest are the same.\r
-- ASIOChannelInfo channelInfo;\r
-- channelInfo.channel = 0;\r
-- channelInfo.isInput = true;\r
-- if ( info.inputChannels <= 0 ) channelInfo.isInput = false;\r
-- result = ASIOGetChannelInfo( &channelInfo );\r
-- if ( result != ASE_OK ) {\r
-- drivers.removeCurrentDriver();\r
-- errorStream_ << "RtApiAsio::getDeviceInfo: error (" << getAsioErrorString( result ) << ") getting driver channel info (" << driverName << ").";\r
-- errorText_ = errorStream_.str();\r
-- error( RtAudioError::WARNING );\r
-- return info;\r
-- }\r
--\r
-- info.nativeFormats = 0;\r
-- if ( channelInfo.type == ASIOSTInt16MSB || channelInfo.type == ASIOSTInt16LSB )\r
-- info.nativeFormats |= RTAUDIO_SINT16;\r
-- else if ( channelInfo.type == ASIOSTInt32MSB || channelInfo.type == ASIOSTInt32LSB )\r
-- info.nativeFormats |= RTAUDIO_SINT32;\r
-- else if ( channelInfo.type == ASIOSTFloat32MSB || channelInfo.type == ASIOSTFloat32LSB )\r
-- info.nativeFormats |= RTAUDIO_FLOAT32;\r
-- else if ( channelInfo.type == ASIOSTFloat64MSB || channelInfo.type == ASIOSTFloat64LSB )\r
-- info.nativeFormats |= RTAUDIO_FLOAT64;\r
-- else if ( channelInfo.type == ASIOSTInt24MSB || channelInfo.type == ASIOSTInt24LSB )\r
-- info.nativeFormats |= RTAUDIO_SINT24;\r
--\r
-- if ( info.outputChannels > 0 )\r
-- if ( getDefaultOutputDevice() == device ) info.isDefaultOutput = true;\r
-- if ( info.inputChannels > 0 )\r
-- if ( getDefaultInputDevice() == device ) info.isDefaultInput = true;\r
--\r
-- info.probed = true;\r
-- drivers.removeCurrentDriver();\r
-- return info;\r
--}\r
--\r
--static void bufferSwitch( long index, ASIOBool /*processNow*/ )\r
--{\r
-- RtApiAsio *object = (RtApiAsio *) asioCallbackInfo->object;\r
-- object->callbackEvent( index );\r
--}\r
--\r
--void RtApiAsio :: saveDeviceInfo( void )\r
--{\r
-- devices_.clear();\r
--\r
-- unsigned int nDevices = getDeviceCount();\r
-- devices_.resize( nDevices );\r
-- for ( unsigned int i=0; i<nDevices; i++ )\r
-- devices_[i] = getDeviceInfo( i );\r
--}\r
--\r
--bool RtApiAsio :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,\r
-- unsigned int firstChannel, unsigned int sampleRate,\r
-- RtAudioFormat format, unsigned int *bufferSize,\r
-- RtAudio::StreamOptions *options )\r
--{////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////\r
--\r
-- bool isDuplexInput = mode == INPUT && stream_.mode == OUTPUT;\r
--\r
-- // For ASIO, a duplex stream MUST use the same driver.\r
-- if ( isDuplexInput && stream_.device[0] != device ) {\r
-- errorText_ = "RtApiAsio::probeDeviceOpen: an ASIO duplex stream must use the same device for input and output!";\r
-- return FAILURE;\r
-- }\r
--\r
-- char driverName[32];\r
-- ASIOError result = drivers.asioGetDriverName( (int) device, driverName, 32 );\r
-- if ( result != ASE_OK ) {\r
-- errorStream_ << "RtApiAsio::probeDeviceOpen: unable to get driver name (" << getAsioErrorString( result ) << ").";\r
-- errorText_ = errorStream_.str();\r
-- return FAILURE;\r
-- }\r
--\r
-- // Only load the driver once for duplex stream.\r
-- if ( !isDuplexInput ) {\r
-- // The getDeviceInfo() function will not work when a stream is open\r
-- // because ASIO does not allow multiple devices to run at the same\r
-- // time. Thus, we'll probe the system before opening a stream and\r
-- // save the results for use by getDeviceInfo().\r
-- this->saveDeviceInfo();\r
--\r
-- if ( !drivers.loadDriver( driverName ) ) {\r
-- errorStream_ << "RtApiAsio::probeDeviceOpen: unable to load driver (" << driverName << ").";\r
-- errorText_ = errorStream_.str();\r
-- return FAILURE;\r
-- }\r
--\r
-- result = ASIOInit( &driverInfo );\r
-- if ( result != ASE_OK ) {\r
-- errorStream_ << "RtApiAsio::probeDeviceOpen: error (" << getAsioErrorString( result ) << ") initializing driver (" << driverName << ").";\r
-- errorText_ = errorStream_.str();\r
-- return FAILURE;\r
-- }\r
-- }\r
--\r
-- // keep them before any "goto error", they are used for error cleanup + goto device boundary checks\r
-- bool buffersAllocated = false;\r
-- AsioHandle *handle = (AsioHandle *) stream_.apiHandle;\r
-- unsigned int nChannels;\r
--\r
--\r
-- // Check the device channel count.\r
-- long inputChannels, outputChannels;\r
-- result = ASIOGetChannels( &inputChannels, &outputChannels );\r
-- if ( result != ASE_OK ) {\r
-- errorStream_ << "RtApiAsio::probeDeviceOpen: error (" << getAsioErrorString( result ) << ") getting channel count (" << driverName << ").";\r
-- errorText_ = errorStream_.str();\r
-- goto error;\r
-- }\r
--\r
-- if ( ( mode == OUTPUT && (channels+firstChannel) > (unsigned int) outputChannels) ||\r
-- ( mode == INPUT && (channels+firstChannel) > (unsigned int) inputChannels) ) {\r
-- errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") does not support requested channel count (" << channels << ") + offset (" << firstChannel << ").";\r
-- errorText_ = errorStream_.str();\r
-- goto error;\r
-- }\r
-- stream_.nDeviceChannels[mode] = channels;\r
-- stream_.nUserChannels[mode] = channels;\r
-- stream_.channelOffset[mode] = firstChannel;\r
--\r
-- // Verify the sample rate is supported.\r
-- result = ASIOCanSampleRate( (ASIOSampleRate) sampleRate );\r
-- if ( result != ASE_OK ) {\r
-- errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") does not support requested sample rate (" << sampleRate << ").";\r
-- errorText_ = errorStream_.str();\r
-- goto error;\r
-- }\r
--\r
-- // Get the current sample rate\r
-- ASIOSampleRate currentRate;\r
-- result = ASIOGetSampleRate( ¤tRate );\r
-- if ( result != ASE_OK ) {\r
-- errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error getting sample rate.";\r
-- errorText_ = errorStream_.str();\r
-- goto error;\r
-- }\r
--\r
-- // Set the sample rate only if necessary\r
-- if ( currentRate != sampleRate ) {\r
-- result = ASIOSetSampleRate( (ASIOSampleRate) sampleRate );\r
-- if ( result != ASE_OK ) {\r
-- errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error setting sample rate (" << sampleRate << ").";\r
-- errorText_ = errorStream_.str();\r
-- goto error;\r
-- }\r
-- }\r
--\r
-- // Determine the driver data type.\r
-- ASIOChannelInfo channelInfo;\r
-- channelInfo.channel = 0;\r
-- if ( mode == OUTPUT ) channelInfo.isInput = false;\r
-- else channelInfo.isInput = true;\r
-- result = ASIOGetChannelInfo( &channelInfo );\r
-- if ( result != ASE_OK ) {\r
-- errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error (" << getAsioErrorString( result ) << ") getting data format.";\r
-- errorText_ = errorStream_.str();\r
-- goto error;\r
-- }\r
--\r
-- // Assuming WINDOWS host is always little-endian.\r
-- stream_.doByteSwap[mode] = false;\r
-- stream_.userFormat = format;\r
-- stream_.deviceFormat[mode] = 0;\r
-- if ( channelInfo.type == ASIOSTInt16MSB || channelInfo.type == ASIOSTInt16LSB ) {\r
-- stream_.deviceFormat[mode] = RTAUDIO_SINT16;\r
-- if ( channelInfo.type == ASIOSTInt16MSB ) stream_.doByteSwap[mode] = true;\r
-- }\r
-- else if ( channelInfo.type == ASIOSTInt32MSB || channelInfo.type == ASIOSTInt32LSB ) {\r
-- stream_.deviceFormat[mode] = RTAUDIO_SINT32;\r
-- if ( channelInfo.type == ASIOSTInt32MSB ) stream_.doByteSwap[mode] = true;\r
-- }\r
-- else if ( channelInfo.type == ASIOSTFloat32MSB || channelInfo.type == ASIOSTFloat32LSB ) {\r
-- stream_.deviceFormat[mode] = RTAUDIO_FLOAT32;\r
-- if ( channelInfo.type == ASIOSTFloat32MSB ) stream_.doByteSwap[mode] = true;\r
-- }\r
-- else if ( channelInfo.type == ASIOSTFloat64MSB || channelInfo.type == ASIOSTFloat64LSB ) {\r
-- stream_.deviceFormat[mode] = RTAUDIO_FLOAT64;\r
-- if ( channelInfo.type == ASIOSTFloat64MSB ) stream_.doByteSwap[mode] = true;\r
-- }\r
-- else if ( channelInfo.type == ASIOSTInt24MSB || channelInfo.type == ASIOSTInt24LSB ) {\r
-- stream_.deviceFormat[mode] = RTAUDIO_SINT24;\r
-- if ( channelInfo.type == ASIOSTInt24MSB ) stream_.doByteSwap[mode] = true;\r
-- }\r
--\r
-- if ( stream_.deviceFormat[mode] == 0 ) {\r
-- errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") data format not supported by RtAudio.";\r
-- errorText_ = errorStream_.str();\r
-- goto error;\r
-- }\r
--\r
-- // Set the buffer size. For a duplex stream, this will end up\r
-- // setting the buffer size based on the input constraints, which\r
-- // should be ok.\r
-- long minSize, maxSize, preferSize, granularity;\r
-- result = ASIOGetBufferSize( &minSize, &maxSize, &preferSize, &granularity );\r
-- if ( result != ASE_OK ) {\r
-- errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error (" << getAsioErrorString( result ) << ") getting buffer size.";\r
-- errorText_ = errorStream_.str();\r
-- goto error;\r
-- }\r
--\r
-- if ( isDuplexInput ) {\r
-- // When this is the duplex input (output was opened before), then we have to use the same\r
-- // buffersize as the output, because it might use the preferred buffer size, which most\r
-- // likely wasn't passed as input to this. The buffer sizes have to be identically anyway,\r
-- // So instead of throwing an error, make them equal. The caller uses the reference\r
-- // to the "bufferSize" param as usual to set up processing buffers.\r
--\r
-- *bufferSize = stream_.bufferSize;\r
--\r
-- } else {\r
-- if ( *bufferSize == 0 ) *bufferSize = preferSize;\r
-- else if ( *bufferSize < (unsigned int) minSize ) *bufferSize = (unsigned int) minSize;\r
-- else if ( *bufferSize > (unsigned int) maxSize ) *bufferSize = (unsigned int) maxSize;\r
-- else if ( granularity == -1 ) {\r
-- // Make sure bufferSize is a power of two.\r
-- int log2_of_min_size = 0;\r
-- int log2_of_max_size = 0;\r
--\r
-- for ( unsigned int i = 0; i < sizeof(long) * 8; i++ ) {\r
-- if ( minSize & ((long)1 << i) ) log2_of_min_size = i;\r
-- if ( maxSize & ((long)1 << i) ) log2_of_max_size = i;\r
-- }\r
--\r
-- long min_delta = std::abs( (long)*bufferSize - ((long)1 << log2_of_min_size) );\r
-- int min_delta_num = log2_of_min_size;\r
--\r
-- for (int i = log2_of_min_size + 1; i <= log2_of_max_size; i++) {\r
-- long current_delta = std::abs( (long)*bufferSize - ((long)1 << i) );\r
-- if (current_delta < min_delta) {\r
-- min_delta = current_delta;\r
-- min_delta_num = i;\r
-- }\r
-- }\r
--\r
-- *bufferSize = ( (unsigned int)1 << min_delta_num );\r
-- if ( *bufferSize < (unsigned int) minSize ) *bufferSize = (unsigned int) minSize;\r
-- else if ( *bufferSize > (unsigned int) maxSize ) *bufferSize = (unsigned int) maxSize;\r
-- }\r
-- else if ( granularity != 0 ) {\r
-- // Set to an even multiple of granularity, rounding up.\r
-- *bufferSize = (*bufferSize + granularity-1) / granularity * granularity;\r
-- }\r
-- }\r
--\r
-- /*\r
-- // we don't use it anymore, see above!\r
-- // Just left it here for the case...\r
-- if ( isDuplexInput && stream_.bufferSize != *bufferSize ) {\r
-- errorText_ = "RtApiAsio::probeDeviceOpen: input/output buffersize discrepancy!";\r
-- goto error;\r
-- }\r
-- */\r
--\r
-- stream_.bufferSize = *bufferSize;\r
-- stream_.nBuffers = 2;\r
--\r
-- if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) stream_.userInterleaved = false;\r
-- else stream_.userInterleaved = true;\r
--\r
-- // ASIO always uses non-interleaved buffers.\r
-- stream_.deviceInterleaved[mode] = false;\r
--\r
-- // Allocate, if necessary, our AsioHandle structure for the stream.\r
-- if ( handle == 0 ) {\r
-- try {\r
-- handle = new AsioHandle;\r
-- }\r
-- catch ( std::bad_alloc& ) {\r
-- errorText_ = "RtApiAsio::probeDeviceOpen: error allocating AsioHandle memory.";\r
-- goto error;\r
-- }\r
-- handle->bufferInfos = 0;\r
--\r
-- // Create a manual-reset event.\r
-- handle->condition = CreateEvent( NULL, // no security\r
-- TRUE, // manual-reset\r
-- FALSE, // non-signaled initially\r
-- NULL ); // unnamed\r
-- stream_.apiHandle = (void *) handle;\r
-- }\r
--\r
-- // Create the ASIO internal buffers. Since RtAudio sets up input\r
-- // and output separately, we'll have to dispose of previously\r
-- // created output buffers for a duplex stream.\r
-- if ( mode == INPUT && stream_.mode == OUTPUT ) {\r
-- ASIODisposeBuffers();\r
-- if ( handle->bufferInfos ) free( handle->bufferInfos );\r
-- }\r
--\r
-- // Allocate, initialize, and save the bufferInfos in our stream callbackInfo structure.\r
-- unsigned int i;\r
-- nChannels = stream_.nDeviceChannels[0] + stream_.nDeviceChannels[1];\r
-- handle->bufferInfos = (ASIOBufferInfo *) malloc( nChannels * sizeof(ASIOBufferInfo) );\r
-- if ( handle->bufferInfos == NULL ) {\r
-- errorStream_ << "RtApiAsio::probeDeviceOpen: error allocating bufferInfo memory for driver (" << driverName << ").";\r
-- errorText_ = errorStream_.str();\r
-- goto error;\r
-- }\r
--\r
-- ASIOBufferInfo *infos;\r
-- infos = handle->bufferInfos;\r
-- for ( i=0; i<stream_.nDeviceChannels[0]; i++, infos++ ) {\r
-- infos->isInput = ASIOFalse;\r
-- infos->channelNum = i + stream_.channelOffset[0];\r
-- infos->buffers[0] = infos->buffers[1] = 0;\r
-- }\r
-- for ( i=0; i<stream_.nDeviceChannels[1]; i++, infos++ ) {\r
-- infos->isInput = ASIOTrue;\r
-- infos->channelNum = i + stream_.channelOffset[1];\r
-- infos->buffers[0] = infos->buffers[1] = 0;\r
-- }\r
--\r
-- // prepare for callbacks\r
-- stream_.sampleRate = sampleRate;\r
-- stream_.device[mode] = device;\r
-- stream_.mode = isDuplexInput ? DUPLEX : mode;\r
--\r
-- // store this class instance before registering callbacks, that are going to use it\r
-- asioCallbackInfo = &stream_.callbackInfo;\r
-- stream_.callbackInfo.object = (void *) this;\r
--\r
-- // Set up the ASIO callback structure and create the ASIO data buffers.\r
-- asioCallbacks.bufferSwitch = &bufferSwitch;\r
-- asioCallbacks.sampleRateDidChange = &sampleRateChanged;\r
-- asioCallbacks.asioMessage = &asioMessages;\r
-- asioCallbacks.bufferSwitchTimeInfo = NULL;\r
-- result = ASIOCreateBuffers( handle->bufferInfos, nChannels, stream_.bufferSize, &asioCallbacks );\r
-- if ( result != ASE_OK ) {\r
-- // Standard method failed. This can happen with strict/misbehaving drivers that return valid buffer size ranges\r
-- // but only accept the preferred buffer size as parameter for ASIOCreateBuffers. eg. Creatives ASIO driver\r
-- // in that case, let's be naïve and try that instead\r
-- *bufferSize = preferSize;\r
-- stream_.bufferSize = *bufferSize;\r
-- result = ASIOCreateBuffers( handle->bufferInfos, nChannels, stream_.bufferSize, &asioCallbacks );\r
-- }\r
--\r
-- if ( result != ASE_OK ) {\r
-- errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error (" << getAsioErrorString( result ) << ") creating buffers.";\r
-- errorText_ = errorStream_.str();\r
-- goto error;\r
-- }\r
-- buffersAllocated = true;\r
-- stream_.state = STREAM_STOPPED;\r
--\r
-- // Set flags for buffer conversion.\r
-- stream_.doConvertBuffer[mode] = false;\r
-- if ( stream_.userFormat != stream_.deviceFormat[mode] )\r
-- stream_.doConvertBuffer[mode] = true;\r
-- if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] &&\r
-- stream_.nUserChannels[mode] > 1 )\r
-- stream_.doConvertBuffer[mode] = true;\r
--\r
-- // Allocate necessary internal buffers\r
-- unsigned long bufferBytes;\r
-- bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat );\r
-- stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 );\r
-- if ( stream_.userBuffer[mode] == NULL ) {\r
-- errorText_ = "RtApiAsio::probeDeviceOpen: error allocating user buffer memory.";\r
-- goto error;\r
-- }\r
--\r
-- if ( stream_.doConvertBuffer[mode] ) {\r
--\r
-- bool makeBuffer = true;\r
-- bufferBytes = stream_.nDeviceChannels[mode] * formatBytes( stream_.deviceFormat[mode] );\r
-- if ( isDuplexInput && stream_.deviceBuffer ) {\r
-- unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] );\r
-- if ( bufferBytes <= bytesOut ) makeBuffer = false;\r
-- }\r
--\r
-- if ( makeBuffer ) {\r
-- bufferBytes *= *bufferSize;\r
-- if ( stream_.deviceBuffer ) free( stream_.deviceBuffer );\r
-- stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 );\r
-- if ( stream_.deviceBuffer == NULL ) {\r
-- errorText_ = "RtApiAsio::probeDeviceOpen: error allocating device buffer memory.";\r
-- goto error;\r
-- }\r
-- }\r
-- }\r
--\r
-- // Determine device latencies\r
-- long inputLatency, outputLatency;\r
-- result = ASIOGetLatencies( &inputLatency, &outputLatency );\r
-- if ( result != ASE_OK ) {\r
-- errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error (" << getAsioErrorString( result ) << ") getting latency.";\r
-- errorText_ = errorStream_.str();\r
-- error( RtAudioError::WARNING); // warn but don't fail\r
-- }\r
-- else {\r
-- stream_.latency[0] = outputLatency;\r
-- stream_.latency[1] = inputLatency;\r
-- }\r
--\r
-- // Setup the buffer conversion information structure. We don't use\r
-- // buffers to do channel offsets, so we override that parameter\r
-- // here.\r
-- if ( stream_.doConvertBuffer[mode] ) setConvertInfo( mode, 0 );\r
--\r
-- return SUCCESS;\r
--\r
-- error:\r
-- if ( !isDuplexInput ) {\r
-- // the cleanup for error in the duplex input, is done by RtApi::openStream\r
-- // So we clean up for single channel only\r
--\r
-- if ( buffersAllocated )\r
-- ASIODisposeBuffers();\r
--\r
-- drivers.removeCurrentDriver();\r
--\r
-- if ( handle ) {\r
-- CloseHandle( handle->condition );\r
-- if ( handle->bufferInfos )\r
-- free( handle->bufferInfos );\r
--\r
-- delete handle;\r
-- stream_.apiHandle = 0;\r
-- }\r
--\r
--\r
-- if ( stream_.userBuffer[mode] ) {\r
-- free( stream_.userBuffer[mode] );\r
-- stream_.userBuffer[mode] = 0;\r
-- }\r
--\r
-- if ( stream_.deviceBuffer ) {\r
-- free( stream_.deviceBuffer );\r
-- stream_.deviceBuffer = 0;\r
-- }\r
-- }\r
--\r
-- return FAILURE;\r
--}////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////\r
--\r
--void RtApiAsio :: closeStream()\r
--{\r
-- if ( stream_.state == STREAM_CLOSED ) {\r
-- errorText_ = "RtApiAsio::closeStream(): no open stream to close!";\r
-- error( RtAudioError::WARNING );\r
-- return;\r
-- }\r
--\r
-- if ( stream_.state == STREAM_RUNNING ) {\r
-- stream_.state = STREAM_STOPPED;\r
-- ASIOStop();\r
-- }\r
-- ASIODisposeBuffers();\r
-- drivers.removeCurrentDriver();\r
--\r
-- AsioHandle *handle = (AsioHandle *) stream_.apiHandle;\r
-- if ( handle ) {\r
-- CloseHandle( handle->condition );\r
-- if ( handle->bufferInfos )\r
-- free( handle->bufferInfos );\r
-- delete handle;\r
-- stream_.apiHandle = 0;\r
-- }\r
--\r
-- for ( int i=0; i<2; i++ ) {\r
-- if ( stream_.userBuffer[i] ) {\r
-- free( stream_.userBuffer[i] );\r
-- stream_.userBuffer[i] = 0;\r
-- }\r
-- }\r
--\r
-- if ( stream_.deviceBuffer ) {\r
-- free( stream_.deviceBuffer );\r
-- stream_.deviceBuffer = 0;\r
-- }\r
--\r
-- stream_.mode = UNINITIALIZED;\r
-- stream_.state = STREAM_CLOSED;\r
--}\r
--\r
--bool stopThreadCalled = false;\r
--\r
--void RtApiAsio :: startStream()\r
--{\r
-- verifyStream();\r
-- if ( stream_.state == STREAM_RUNNING ) {\r
-- errorText_ = "RtApiAsio::startStream(): the stream is already running!";\r
-- error( RtAudioError::WARNING );\r
-- return;\r
-- }\r
--\r
-- AsioHandle *handle = (AsioHandle *) stream_.apiHandle;\r
-- ASIOError result = ASIOStart();\r
-- if ( result != ASE_OK ) {\r
-- errorStream_ << "RtApiAsio::startStream: error (" << getAsioErrorString( result ) << ") starting device.";\r
-- errorText_ = errorStream_.str();\r
-- goto unlock;\r
-- }\r
--\r
-- handle->drainCounter = 0;\r
-- handle->internalDrain = false;\r
-- ResetEvent( handle->condition );\r
-- stream_.state = STREAM_RUNNING;\r
-- asioXRun = false;\r
--\r
-- unlock:\r
-- stopThreadCalled = false;\r
--\r
-- if ( result == ASE_OK ) return;\r
-- error( RtAudioError::SYSTEM_ERROR );\r
--}\r
--\r
--void RtApiAsio :: stopStream()\r
--{\r
-- verifyStream();\r
-- if ( stream_.state == STREAM_STOPPED ) {\r
-- errorText_ = "RtApiAsio::stopStream(): the stream is already stopped!";\r
-- error( RtAudioError::WARNING );\r
-- return;\r
-- }\r
--\r
-- AsioHandle *handle = (AsioHandle *) stream_.apiHandle;\r
-- if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {\r
-- if ( handle->drainCounter == 0 ) {\r
-- handle->drainCounter = 2;\r
-- WaitForSingleObject( handle->condition, INFINITE ); // block until signaled\r
-- }\r
-- }\r
--\r
-- stream_.state = STREAM_STOPPED;\r
--\r
-- ASIOError result = ASIOStop();\r
-- if ( result != ASE_OK ) {\r
-- errorStream_ << "RtApiAsio::stopStream: error (" << getAsioErrorString( result ) << ") stopping device.";\r
-- errorText_ = errorStream_.str();\r
-- }\r
--\r
-- if ( result == ASE_OK ) return;\r
-- error( RtAudioError::SYSTEM_ERROR );\r
--}\r
--\r
--void RtApiAsio :: abortStream()\r
--{\r
-- verifyStream();\r
-- if ( stream_.state == STREAM_STOPPED ) {\r
-- errorText_ = "RtApiAsio::abortStream(): the stream is already stopped!";\r
-- error( RtAudioError::WARNING );\r
-- return;\r
-- }\r
--\r
-- // The following lines were commented-out because some behavior was\r
-- // noted where the device buffers need to be zeroed to avoid\r
-- // continuing sound, even when the device buffers are completely\r
-- // disposed. So now, calling abort is the same as calling stop.\r
-- // AsioHandle *handle = (AsioHandle *) stream_.apiHandle;\r
-- // handle->drainCounter = 2;\r
-- stopStream();\r
--}\r
--\r
--// This function will be called by a spawned thread when the user\r
--// callback function signals that the stream should be stopped or\r
--// aborted. It is necessary to handle it this way because the\r
--// callbackEvent() function must return before the ASIOStop()\r
--// function will return.\r
--static unsigned __stdcall asioStopStream( void *ptr )\r
--{\r
-- CallbackInfo *info = (CallbackInfo *) ptr;\r
-- RtApiAsio *object = (RtApiAsio *) info->object;\r
--\r
-- object->stopStream();\r
-- _endthreadex( 0 );\r
-- return 0;\r
--}\r
--\r
--bool RtApiAsio :: callbackEvent( long bufferIndex )\r
--{\r
-- if ( stream_.state == STREAM_STOPPED || stream_.state == STREAM_STOPPING ) return SUCCESS;\r
-- if ( stream_.state == STREAM_CLOSED ) {\r
-- errorText_ = "RtApiAsio::callbackEvent(): the stream is closed ... this shouldn't happen!";\r
-- error( RtAudioError::WARNING );\r
-- return FAILURE;\r
-- }\r
--\r
-- CallbackInfo *info = (CallbackInfo *) &stream_.callbackInfo;\r
-- AsioHandle *handle = (AsioHandle *) stream_.apiHandle;\r
--\r
-- // Check if we were draining the stream and signal if finished.\r
-- if ( handle->drainCounter > 3 ) {\r
--\r
-- stream_.state = STREAM_STOPPING;\r
-- if ( handle->internalDrain == false )\r
-- SetEvent( handle->condition );\r
-- else { // spawn a thread to stop the stream\r
-- unsigned threadId;\r
-- stream_.callbackInfo.thread = _beginthreadex( NULL, 0, &asioStopStream,\r
-- &stream_.callbackInfo, 0, &threadId );\r
-- }\r
-- return SUCCESS;\r
-- }\r
--\r
-- // Invoke user callback to get fresh output data UNLESS we are\r
-- // draining stream.\r
-- if ( handle->drainCounter == 0 ) {\r
-- RtAudioCallback callback = (RtAudioCallback) info->callback;\r
-- double streamTime = getStreamTime();\r
-- RtAudioStreamStatus status = 0;\r
-- if ( stream_.mode != INPUT && asioXRun == true ) {\r
-- status |= RTAUDIO_OUTPUT_UNDERFLOW;\r
-- asioXRun = false;\r
-- }\r
-- if ( stream_.mode != OUTPUT && asioXRun == true ) {\r
-- status |= RTAUDIO_INPUT_OVERFLOW;\r
-- asioXRun = false;\r
-- }\r
-- int cbReturnValue = callback( stream_.userBuffer[0], stream_.userBuffer[1],\r
-- stream_.bufferSize, streamTime, status, info->userData );\r
-- if ( cbReturnValue == 2 ) {\r
-- stream_.state = STREAM_STOPPING;\r
-- handle->drainCounter = 2;\r
-- unsigned threadId;\r
-- stream_.callbackInfo.thread = _beginthreadex( NULL, 0, &asioStopStream,\r
-- &stream_.callbackInfo, 0, &threadId );\r
-- return SUCCESS;\r
-- }\r
-- else if ( cbReturnValue == 1 ) {\r
-- handle->drainCounter = 1;\r
-- handle->internalDrain = true;\r
-- }\r
-- }\r
--\r
-- unsigned int nChannels, bufferBytes, i, j;\r
-- nChannels = stream_.nDeviceChannels[0] + stream_.nDeviceChannels[1];\r
-- if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {\r
--\r
-- bufferBytes = stream_.bufferSize * formatBytes( stream_.deviceFormat[0] );\r
--\r
-- if ( handle->drainCounter > 1 ) { // write zeros to the output stream\r
--\r
-- for ( i=0, j=0; i<nChannels; i++ ) {\r
-- if ( handle->bufferInfos[i].isInput != ASIOTrue )\r
-- memset( handle->bufferInfos[i].buffers[bufferIndex], 0, bufferBytes );\r
-- }\r
--\r
-- }\r
-- else if ( stream_.doConvertBuffer[0] ) {\r
--\r
-- convertBuffer( stream_.deviceBuffer, stream_.userBuffer[0], stream_.convertInfo[0] );\r
-- if ( stream_.doByteSwap[0] )\r
-- byteSwapBuffer( stream_.deviceBuffer,\r
-- stream_.bufferSize * stream_.nDeviceChannels[0],\r
-- stream_.deviceFormat[0] );\r
--\r
-- for ( i=0, j=0; i<nChannels; i++ ) {\r
-- if ( handle->bufferInfos[i].isInput != ASIOTrue )\r
-- memcpy( handle->bufferInfos[i].buffers[bufferIndex],\r
-- &stream_.deviceBuffer[j++*bufferBytes], bufferBytes );\r
-- }\r
--\r
-- }\r
-- else {\r
--\r
-- if ( stream_.doByteSwap[0] )\r
-- byteSwapBuffer( stream_.userBuffer[0],\r
-- stream_.bufferSize * stream_.nUserChannels[0],\r
-- stream_.userFormat );\r
--\r
-- for ( i=0, j=0; i<nChannels; i++ ) {\r
-- if ( handle->bufferInfos[i].isInput != ASIOTrue )\r
-- memcpy( handle->bufferInfos[i].buffers[bufferIndex],\r
-- &stream_.userBuffer[0][bufferBytes*j++], bufferBytes );\r
-- }\r
--\r
-- }\r
-- }\r
--\r
-- // Don't bother draining input\r
-- if ( handle->drainCounter ) {\r
-- handle->drainCounter++;\r
-- goto unlock;\r
-- }\r
--\r
-- if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {\r
--\r
-- bufferBytes = stream_.bufferSize * formatBytes(stream_.deviceFormat[1]);\r
--\r
-- if (stream_.doConvertBuffer[1]) {\r
--\r
-- // Always interleave ASIO input data.\r
-- for ( i=0, j=0; i<nChannels; i++ ) {\r
-- if ( handle->bufferInfos[i].isInput == ASIOTrue )\r
-- memcpy( &stream_.deviceBuffer[j++*bufferBytes],\r
-- handle->bufferInfos[i].buffers[bufferIndex],\r
-- bufferBytes );\r
-- }\r
--\r
-- if ( stream_.doByteSwap[1] )\r
-- byteSwapBuffer( stream_.deviceBuffer,\r
-- stream_.bufferSize * stream_.nDeviceChannels[1],\r
-- stream_.deviceFormat[1] );\r
-- convertBuffer( stream_.userBuffer[1], stream_.deviceBuffer, stream_.convertInfo[1] );\r
--\r
-- }\r
-- else {\r
-- for ( i=0, j=0; i<nChannels; i++ ) {\r
-- if ( handle->bufferInfos[i].isInput == ASIOTrue ) {\r
-- memcpy( &stream_.userBuffer[1][bufferBytes*j++],\r
-- handle->bufferInfos[i].buffers[bufferIndex],\r
-- bufferBytes );\r
-- }\r
-- }\r
--\r
-- if ( stream_.doByteSwap[1] )\r
-- byteSwapBuffer( stream_.userBuffer[1],\r
-- stream_.bufferSize * stream_.nUserChannels[1],\r
-- stream_.userFormat );\r
-- }\r
-- }\r
--\r
-- unlock:\r
-- // The following call was suggested by Malte Clasen. While the API\r
-- // documentation indicates it should not be required, some device\r
-- // drivers apparently do not function correctly without it.\r
-- ASIOOutputReady();\r
--\r
-- RtApi::tickStreamTime();\r
-- return SUCCESS;\r
--}\r
--\r
--static void sampleRateChanged( ASIOSampleRate sRate )\r
--{\r
-- // The ASIO documentation says that this usually only happens during\r
-- // external sync. Audio processing is not stopped by the driver,\r
-- // actual sample rate might not have even changed, maybe only the\r
-- // sample rate status of an AES/EBU or S/PDIF digital input at the\r
-- // audio device.\r
--\r
-- RtApi *object = (RtApi *) asioCallbackInfo->object;\r
-- try {\r
-- object->stopStream();\r
-- }\r
-- catch ( RtAudioError &exception ) {\r
-- std::cerr << "\nRtApiAsio: sampleRateChanged() error (" << exception.getMessage() << ")!\n" << std::endl;\r
-- return;\r
-- }\r
--\r
-- std::cerr << "\nRtApiAsio: driver reports sample rate changed to " << sRate << " ... stream stopped!!!\n" << std::endl;\r
--}\r
--\r
--static long asioMessages( long selector, long value, void* /*message*/, double* /*opt*/ )\r
--{\r
-- long ret = 0;\r
--\r
-- switch( selector ) {\r
-- case kAsioSelectorSupported:\r
-- if ( value == kAsioResetRequest\r
-- || value == kAsioEngineVersion\r
-- || value == kAsioResyncRequest\r
-- || value == kAsioLatenciesChanged\r
-- // The following three were added for ASIO 2.0, you don't\r
-- // necessarily have to support them.\r
-- || value == kAsioSupportsTimeInfo\r
-- || value == kAsioSupportsTimeCode\r
-- || value == kAsioSupportsInputMonitor)\r
-- ret = 1L;\r
-- break;\r
-- case kAsioResetRequest:\r
-- // Defer the task and perform the reset of the driver during the\r
-- // next "safe" situation. You cannot reset the driver right now,\r
-- // as this code is called from the driver. Reset the driver is\r
-- // done by completely destruct is. I.e. ASIOStop(),\r
-- // ASIODisposeBuffers(), Destruction Afterwards you initialize the\r
-- // driver again.\r
-- std::cerr << "\nRtApiAsio: driver reset requested!!!" << std::endl;\r
-- ret = 1L;\r
-- break;\r
-- case kAsioResyncRequest:\r
-- // This informs the application that the driver encountered some\r
-- // non-fatal data loss. It is used for synchronization purposes\r
-- // of different media. Added mainly to work around the Win16Mutex\r
-- // problems in Windows 95/98 with the Windows Multimedia system,\r
-- // which could lose data because the Mutex was held too long by\r
-- // another thread. However a driver can issue it in other\r
-- // situations, too.\r
-- // std::cerr << "\nRtApiAsio: driver resync requested!!!" << std::endl;\r
-- asioXRun = true;\r
-- ret = 1L;\r
-- break;\r
-- case kAsioLatenciesChanged:\r
-- // This will inform the host application that the drivers were\r
-- // latencies changed. Beware, it this does not mean that the\r
-- // buffer sizes have changed! You might need to update internal\r
-- // delay data.\r
-- std::cerr << "\nRtApiAsio: driver latency may have changed!!!" << std::endl;\r
-- ret = 1L;\r
-- break;\r
-- case kAsioEngineVersion:\r
-- // Return the supported ASIO version of the host application. If\r
-- // a host application does not implement this selector, ASIO 1.0\r
-- // is assumed by the driver.\r
-- ret = 2L;\r
-- break;\r
-- case kAsioSupportsTimeInfo:\r
-- // Informs the driver whether the\r
-- // asioCallbacks.bufferSwitchTimeInfo() callback is supported.\r
-- // For compatibility with ASIO 1.0 drivers the host application\r
-- // should always support the "old" bufferSwitch method, too.\r
-- ret = 0;\r
-- break;\r
-- case kAsioSupportsTimeCode:\r
-- // Informs the driver whether application is interested in time\r
-- // code info. If an application does not need to know about time\r
-- // code, the driver has less work to do.\r
-- ret = 0;\r
-- break;\r
-- }\r
-- return ret;\r
--}\r
--\r
--static const char* getAsioErrorString( ASIOError result )\r
--{\r
-- struct Messages\r
-- {\r
-- ASIOError value;\r
-- const char*message;\r
-- };\r
--\r
-- static const Messages m[] =\r
-- {\r
-- { ASE_NotPresent, "Hardware input or output is not present or available." },\r
-- { ASE_HWMalfunction, "Hardware is malfunctioning." },\r
-- { ASE_InvalidParameter, "Invalid input parameter." },\r
-- { ASE_InvalidMode, "Invalid mode." },\r
-- { ASE_SPNotAdvancing, "Sample position not advancing." },\r
-- { ASE_NoClock, "Sample clock or rate cannot be determined or is not present." },\r
-- { ASE_NoMemory, "Not enough memory to complete the request." }\r
-- };\r
--\r
-- for ( unsigned int i = 0; i < sizeof(m)/sizeof(m[0]); ++i )\r
-- if ( m[i].value == result ) return m[i].message;\r
--\r
-- return "Unknown error.";\r
--}\r
--\r
--//******************** End of __WINDOWS_ASIO__ *********************//\r
--#endif\r
--\r
--\r
--#if defined(__WINDOWS_WASAPI__) // Windows WASAPI API\r
--\r
--// Authored by Marcus Tomlinson <themarcustomlinson@gmail.com>, April 2014\r
--// - Introduces support for the Windows WASAPI API\r
--// - Aims to deliver bit streams to and from hardware at the lowest possible latency, via the absolute minimum buffer sizes required\r
--// - Provides flexible stream configuration to an otherwise strict and inflexible WASAPI interface\r
--// - Includes automatic internal conversion of sample rate and buffer size between hardware and the user\r
--\r
--#ifndef INITGUID\r
-- #define INITGUID\r
--#endif\r
--#include <audioclient.h>\r
--#include <avrt.h>\r
--#include <mmdeviceapi.h>\r
--#include <functiondiscoverykeys_devpkey.h>\r
--\r
--//=============================================================================\r
--\r
--#define SAFE_RELEASE( objectPtr )\\r
--if ( objectPtr )\\r
--{\\r
-- objectPtr->Release();\\r
-- objectPtr = NULL;\\r
--}\r
--\r
--typedef HANDLE ( __stdcall *TAvSetMmThreadCharacteristicsPtr )( LPCWSTR TaskName, LPDWORD TaskIndex );\r
--\r
--//-----------------------------------------------------------------------------\r
--\r
--// WASAPI dictates stream sample rate, format, channel count, and in some cases, buffer size.\r
--// Therefore we must perform all necessary conversions to user buffers in order to satisfy these\r
--// requirements. WasapiBuffer ring buffers are used between HwIn->UserIn and UserOut->HwOut to\r
--// provide intermediate storage for read / write synchronization.\r
--class WasapiBuffer\r
--{\r
--public:\r
-- WasapiBuffer()\r
-- : buffer_( NULL ),\r
-- bufferSize_( 0 ),\r
-- inIndex_( 0 ),\r
-- outIndex_( 0 ) {}\r
--\r
-- ~WasapiBuffer() {\r
-- free( buffer_ );\r
-- }\r
--\r
-- // sets the length of the internal ring buffer\r
-- void setBufferSize( unsigned int bufferSize, unsigned int formatBytes ) {\r
-- free( buffer_ );\r
--\r
-- buffer_ = ( char* ) calloc( bufferSize, formatBytes );\r
--\r
-- bufferSize_ = bufferSize;\r
-- inIndex_ = 0;\r
-- outIndex_ = 0;\r
-- }\r
--\r
-- // attempt to push a buffer into the ring buffer at the current "in" index\r
-- bool pushBuffer( char* buffer, unsigned int bufferSize, RtAudioFormat format )\r
-- {\r
-- if ( !buffer || // incoming buffer is NULL\r
-- bufferSize == 0 || // incoming buffer has no data\r
-- bufferSize > bufferSize_ ) // incoming buffer too large\r
-- {\r
-- return false;\r
-- }\r
--\r
-- unsigned int relOutIndex = outIndex_;\r
-- unsigned int inIndexEnd = inIndex_ + bufferSize;\r
-- if ( relOutIndex < inIndex_ && inIndexEnd >= bufferSize_ ) {\r
-- relOutIndex += bufferSize_;\r
-- }\r
--\r
-- // "in" index can end on the "out" index but cannot begin at it\r
-- if ( inIndex_ <= relOutIndex && inIndexEnd > relOutIndex ) {\r
-- return false; // not enough space between "in" index and "out" index\r
-- }\r
--\r
-- // copy buffer from external to internal\r
-- int fromZeroSize = inIndex_ + bufferSize - bufferSize_;\r
-- fromZeroSize = fromZeroSize < 0 ? 0 : fromZeroSize;\r
-- int fromInSize = bufferSize - fromZeroSize;\r
--\r
-- switch( format )\r
-- {\r
-- case RTAUDIO_SINT8:\r
-- memcpy( &( ( char* ) buffer_ )[inIndex_], buffer, fromInSize * sizeof( char ) );\r
-- memcpy( buffer_, &( ( char* ) buffer )[fromInSize], fromZeroSize * sizeof( char ) );\r
-- break;\r
-- case RTAUDIO_SINT16:\r
-- memcpy( &( ( short* ) buffer_ )[inIndex_], buffer, fromInSize * sizeof( short ) );\r
-- memcpy( buffer_, &( ( short* ) buffer )[fromInSize], fromZeroSize * sizeof( short ) );\r
-- break;\r
-- case RTAUDIO_SINT24:\r
-- memcpy( &( ( S24* ) buffer_ )[inIndex_], buffer, fromInSize * sizeof( S24 ) );\r
-- memcpy( buffer_, &( ( S24* ) buffer )[fromInSize], fromZeroSize * sizeof( S24 ) );\r
-- break;\r
-- case RTAUDIO_SINT32:\r
-- memcpy( &( ( int* ) buffer_ )[inIndex_], buffer, fromInSize * sizeof( int ) );\r
-- memcpy( buffer_, &( ( int* ) buffer )[fromInSize], fromZeroSize * sizeof( int ) );\r
-- break;\r
-- case RTAUDIO_FLOAT32:\r
-- memcpy( &( ( float* ) buffer_ )[inIndex_], buffer, fromInSize * sizeof( float ) );\r
-- memcpy( buffer_, &( ( float* ) buffer )[fromInSize], fromZeroSize * sizeof( float ) );\r
-- break;\r
-- case RTAUDIO_FLOAT64:\r
-- memcpy( &( ( double* ) buffer_ )[inIndex_], buffer, fromInSize * sizeof( double ) );\r
-- memcpy( buffer_, &( ( double* ) buffer )[fromInSize], fromZeroSize * sizeof( double ) );\r
-- break;\r
-- }\r
--\r
-- // update "in" index\r
-- inIndex_ += bufferSize;\r
-- inIndex_ %= bufferSize_;\r
--\r
-- return true;\r
-- }\r
--\r
-- // attempt to pull a buffer from the ring buffer from the current "out" index\r
-- bool pullBuffer( char* buffer, unsigned int bufferSize, RtAudioFormat format )\r
-- {\r
-- if ( !buffer || // incoming buffer is NULL\r
-- bufferSize == 0 || // incoming buffer has no data\r
-- bufferSize > bufferSize_ ) // incoming buffer too large\r
-- {\r
-- return false;\r
-- }\r
--\r
-- unsigned int relInIndex = inIndex_;\r
-- unsigned int outIndexEnd = outIndex_ + bufferSize;\r
-- if ( relInIndex < outIndex_ && outIndexEnd >= bufferSize_ ) {\r
-- relInIndex += bufferSize_;\r
-- }\r
--\r
-- // "out" index can begin at and end on the "in" index\r
-- if ( outIndex_ < relInIndex && outIndexEnd > relInIndex ) {\r
-- return false; // not enough space between "out" index and "in" index\r
-- }\r
--\r
-- // copy buffer from internal to external\r
-- int fromZeroSize = outIndex_ + bufferSize - bufferSize_;\r
-- fromZeroSize = fromZeroSize < 0 ? 0 : fromZeroSize;\r
-- int fromOutSize = bufferSize - fromZeroSize;\r
--\r
-- switch( format )\r
-- {\r
-- case RTAUDIO_SINT8:\r
-- memcpy( buffer, &( ( char* ) buffer_ )[outIndex_], fromOutSize * sizeof( char ) );\r
-- memcpy( &( ( char* ) buffer )[fromOutSize], buffer_, fromZeroSize * sizeof( char ) );\r
-- break;\r
-- case RTAUDIO_SINT16:\r
-- memcpy( buffer, &( ( short* ) buffer_ )[outIndex_], fromOutSize * sizeof( short ) );\r
-- memcpy( &( ( short* ) buffer )[fromOutSize], buffer_, fromZeroSize * sizeof( short ) );\r
-- break;\r
-- case RTAUDIO_SINT24:\r
-- memcpy( buffer, &( ( S24* ) buffer_ )[outIndex_], fromOutSize * sizeof( S24 ) );\r
-- memcpy( &( ( S24* ) buffer )[fromOutSize], buffer_, fromZeroSize * sizeof( S24 ) );\r
-- break;\r
-- case RTAUDIO_SINT32:\r
-- memcpy( buffer, &( ( int* ) buffer_ )[outIndex_], fromOutSize * sizeof( int ) );\r
-- memcpy( &( ( int* ) buffer )[fromOutSize], buffer_, fromZeroSize * sizeof( int ) );\r
-- break;\r
-- case RTAUDIO_FLOAT32:\r
-- memcpy( buffer, &( ( float* ) buffer_ )[outIndex_], fromOutSize * sizeof( float ) );\r
-- memcpy( &( ( float* ) buffer )[fromOutSize], buffer_, fromZeroSize * sizeof( float ) );\r
-- break;\r
-- case RTAUDIO_FLOAT64:\r
-- memcpy( buffer, &( ( double* ) buffer_ )[outIndex_], fromOutSize * sizeof( double ) );\r
-- memcpy( &( ( double* ) buffer )[fromOutSize], buffer_, fromZeroSize * sizeof( double ) );\r
-- break;\r
-- }\r
--\r
-- // update "out" index\r
-- outIndex_ += bufferSize;\r
-- outIndex_ %= bufferSize_;\r
--\r
-- return true;\r
-- }\r
--\r
--private:\r
-- char* buffer_;\r
-- unsigned int bufferSize_;\r
-- unsigned int inIndex_;\r
-- unsigned int outIndex_;\r
--};\r
--\r
--//-----------------------------------------------------------------------------\r
--\r
--// In order to satisfy WASAPI's buffer requirements, we need a means of converting sample rate\r
--// between HW and the user. The convertBufferWasapi function is used to perform this conversion\r
--// between HwIn->UserIn and UserOut->HwOut during the stream callback loop.\r
--// This sample rate converter favors speed over quality, and works best with conversions between\r
--// one rate and its multiple.\r
--void convertBufferWasapi( char* outBuffer,\r
-- const char* inBuffer,\r
-- const unsigned int& channelCount,\r
-- const unsigned int& inSampleRate,\r
-- const unsigned int& outSampleRate,\r
-- const unsigned int& inSampleCount,\r
-- unsigned int& outSampleCount,\r
-- const RtAudioFormat& format )\r
--{\r
-- // calculate the new outSampleCount and relative sampleStep\r
-- float sampleRatio = ( float ) outSampleRate / inSampleRate;\r
-- float sampleStep = 1.0f / sampleRatio;\r
-- float inSampleFraction = 0.0f;\r
--\r
-- outSampleCount = ( unsigned int ) roundf( inSampleCount * sampleRatio );\r
--\r
-- // frame-by-frame, copy each relative input sample into it's corresponding output sample\r
-- for ( unsigned int outSample = 0; outSample < outSampleCount; outSample++ )\r
-- {\r
-- unsigned int inSample = ( unsigned int ) inSampleFraction;\r
--\r
-- switch ( format )\r
-- {\r
-- case RTAUDIO_SINT8:\r
-- memcpy( &( ( char* ) outBuffer )[ outSample * channelCount ], &( ( char* ) inBuffer )[ inSample * channelCount ], channelCount * sizeof( char ) );\r
-- break;\r
-- case RTAUDIO_SINT16:\r
-- memcpy( &( ( short* ) outBuffer )[ outSample * channelCount ], &( ( short* ) inBuffer )[ inSample * channelCount ], channelCount * sizeof( short ) );\r
-- break;\r
-- case RTAUDIO_SINT24:\r
-- memcpy( &( ( S24* ) outBuffer )[ outSample * channelCount ], &( ( S24* ) inBuffer )[ inSample * channelCount ], channelCount * sizeof( S24 ) );\r
-- break;\r
-- case RTAUDIO_SINT32:\r
-- memcpy( &( ( int* ) outBuffer )[ outSample * channelCount ], &( ( int* ) inBuffer )[ inSample * channelCount ], channelCount * sizeof( int ) );\r
-- break;\r
-- case RTAUDIO_FLOAT32:\r
-- memcpy( &( ( float* ) outBuffer )[ outSample * channelCount ], &( ( float* ) inBuffer )[ inSample * channelCount ], channelCount * sizeof( float ) );\r
-- break;\r
-- case RTAUDIO_FLOAT64:\r
-- memcpy( &( ( double* ) outBuffer )[ outSample * channelCount ], &( ( double* ) inBuffer )[ inSample * channelCount ], channelCount * sizeof( double ) );\r
-- break;\r
-- }\r
--\r
-- // jump to next in sample\r
-- inSampleFraction += sampleStep;\r
-- }\r
--}\r
--\r
--//-----------------------------------------------------------------------------\r
--\r
--// A structure to hold various information related to the WASAPI implementation.\r
--struct WasapiHandle\r
--{\r
-- IAudioClient* captureAudioClient;\r
-- IAudioClient* renderAudioClient;\r
-- IAudioCaptureClient* captureClient;\r
-- IAudioRenderClient* renderClient;\r
-- HANDLE captureEvent;\r
-- HANDLE renderEvent;\r
--\r
-- WasapiHandle()\r
-- : captureAudioClient( NULL ),\r
-- renderAudioClient( NULL ),\r
-- captureClient( NULL ),\r
-- renderClient( NULL ),\r
-- captureEvent( NULL ),\r
-- renderEvent( NULL ) {}\r
--};\r
--\r
--//=============================================================================\r
--\r
--RtApiWasapi::RtApiWasapi()\r
-- : coInitialized_( false ), deviceEnumerator_( NULL )\r
--{\r
-- // WASAPI can run either apartment or multi-threaded\r
-- HRESULT hr = CoInitialize( NULL );\r
-- if ( !FAILED( hr ) )\r
-- coInitialized_ = true;\r
--\r
-- // Instantiate device enumerator\r
-- hr = CoCreateInstance( __uuidof( MMDeviceEnumerator ), NULL,\r
-- CLSCTX_ALL, __uuidof( IMMDeviceEnumerator ),\r
-- ( void** ) &deviceEnumerator_ );\r
--\r
-- if ( FAILED( hr ) ) {\r
-- errorText_ = "RtApiWasapi::RtApiWasapi: Unable to instantiate device enumerator";\r
-- error( RtAudioError::DRIVER_ERROR );\r
-- }\r
--}\r
--\r
--//-----------------------------------------------------------------------------\r
--\r
--RtApiWasapi::~RtApiWasapi()\r
--{\r
-- if ( stream_.state != STREAM_CLOSED )\r
-- closeStream();\r
--\r
-- SAFE_RELEASE( deviceEnumerator_ );\r
--\r
-- // If this object previously called CoInitialize()\r
-- if ( coInitialized_ )\r
-- CoUninitialize();\r
--}\r
--\r
--//=============================================================================\r
--\r
--unsigned int RtApiWasapi::getDeviceCount( void )\r
--{\r
-- unsigned int captureDeviceCount = 0;\r
-- unsigned int renderDeviceCount = 0;\r
--\r
-- IMMDeviceCollection* captureDevices = NULL;\r
-- IMMDeviceCollection* renderDevices = NULL;\r
--\r
-- // Count capture devices\r
-- errorText_.clear();\r
-- HRESULT hr = deviceEnumerator_->EnumAudioEndpoints( eCapture, DEVICE_STATE_ACTIVE, &captureDevices );\r
-- if ( FAILED( hr ) ) {\r
-- errorText_ = "RtApiWasapi::getDeviceCount: Unable to retrieve capture device collection.";\r
-- goto Exit;\r
-- }\r
--\r
-- hr = captureDevices->GetCount( &captureDeviceCount );\r
-- if ( FAILED( hr ) ) {\r
-- errorText_ = "RtApiWasapi::getDeviceCount: Unable to retrieve capture device count.";\r
-- goto Exit;\r
-- }\r
--\r
-- // Count render devices\r
-- hr = deviceEnumerator_->EnumAudioEndpoints( eRender, DEVICE_STATE_ACTIVE, &renderDevices );\r
-- if ( FAILED( hr ) ) {\r
-- errorText_ = "RtApiWasapi::getDeviceCount: Unable to retrieve render device collection.";\r
-- goto Exit;\r
-- }\r
--\r
-- hr = renderDevices->GetCount( &renderDeviceCount );\r
-- if ( FAILED( hr ) ) {\r
-- errorText_ = "RtApiWasapi::getDeviceCount: Unable to retrieve render device count.";\r
-- goto Exit;\r
-- }\r
--\r
--Exit:\r
-- // release all references\r
-- SAFE_RELEASE( captureDevices );\r
-- SAFE_RELEASE( renderDevices );\r
--\r
-- if ( errorText_.empty() )\r
-- return captureDeviceCount + renderDeviceCount;\r
--\r
-- error( RtAudioError::DRIVER_ERROR );\r
-- return 0;\r
--}\r
--\r
--//-----------------------------------------------------------------------------\r
--\r
--RtAudio::DeviceInfo RtApiWasapi::getDeviceInfo( unsigned int device )\r
--{\r
-- RtAudio::DeviceInfo info;\r
-- unsigned int captureDeviceCount = 0;\r
-- unsigned int renderDeviceCount = 0;\r
-- std::string defaultDeviceName;\r
-- bool isCaptureDevice = false;\r
--\r
-- PROPVARIANT deviceNameProp;\r
-- PROPVARIANT defaultDeviceNameProp;\r
--\r
-- IMMDeviceCollection* captureDevices = NULL;\r
-- IMMDeviceCollection* renderDevices = NULL;\r
-- IMMDevice* devicePtr = NULL;\r
-- IMMDevice* defaultDevicePtr = NULL;\r
-- IAudioClient* audioClient = NULL;\r
-- IPropertyStore* devicePropStore = NULL;\r
-- IPropertyStore* defaultDevicePropStore = NULL;\r
--\r
-- WAVEFORMATEX* deviceFormat = NULL;\r
-- WAVEFORMATEX* closestMatchFormat = NULL;\r
--\r
-- // probed\r
-- info.probed = false;\r
--\r
-- // Count capture devices\r
-- errorText_.clear();\r
-- RtAudioError::Type errorType = RtAudioError::DRIVER_ERROR;\r
-- HRESULT hr = deviceEnumerator_->EnumAudioEndpoints( eCapture, DEVICE_STATE_ACTIVE, &captureDevices );\r
-- if ( FAILED( hr ) ) {\r
-- errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve capture device collection.";\r
-- goto Exit;\r
-- }\r
--\r
-- hr = captureDevices->GetCount( &captureDeviceCount );\r
-- if ( FAILED( hr ) ) {\r
-- errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve capture device count.";\r
-- goto Exit;\r
-- }\r
--\r
-- // Count render devices\r
-- hr = deviceEnumerator_->EnumAudioEndpoints( eRender, DEVICE_STATE_ACTIVE, &renderDevices );\r
-- if ( FAILED( hr ) ) {\r
-- errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve render device collection.";\r
-- goto Exit;\r
-- }\r
--\r
-- hr = renderDevices->GetCount( &renderDeviceCount );\r
-- if ( FAILED( hr ) ) {\r
-- errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve render device count.";\r
-- goto Exit;\r
-- }\r
--\r
-- // validate device index\r
-- if ( device >= captureDeviceCount + renderDeviceCount ) {\r
-- errorText_ = "RtApiWasapi::getDeviceInfo: Invalid device index.";\r
-- errorType = RtAudioError::INVALID_USE;\r
-- goto Exit;\r
-- }\r
--\r
-- // determine whether index falls within capture or render devices\r
-- if ( device >= renderDeviceCount ) {\r
-- hr = captureDevices->Item( device - renderDeviceCount, &devicePtr );\r
-- if ( FAILED( hr ) ) {\r
-- errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve capture device handle.";\r
-- goto Exit;\r
-- }\r
-- isCaptureDevice = true;\r
-- }\r
-- else {\r
-- hr = renderDevices->Item( device, &devicePtr );\r
-- if ( FAILED( hr ) ) {\r
-- errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve render device handle.";\r
-- goto Exit;\r
-- }\r
-- isCaptureDevice = false;\r
-- }\r
--\r
-- // get default device name\r
-- if ( isCaptureDevice ) {\r
-- hr = deviceEnumerator_->GetDefaultAudioEndpoint( eCapture, eConsole, &defaultDevicePtr );\r
-- if ( FAILED( hr ) ) {\r
-- errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve default capture device handle.";\r
-- goto Exit;\r
-- }\r
-- }\r
-- else {\r
-- hr = deviceEnumerator_->GetDefaultAudioEndpoint( eRender, eConsole, &defaultDevicePtr );\r
-- if ( FAILED( hr ) ) {\r
-- errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve default render device handle.";\r
-- goto Exit;\r
-- }\r
-- }\r
--\r
-- hr = defaultDevicePtr->OpenPropertyStore( STGM_READ, &defaultDevicePropStore );\r
-- if ( FAILED( hr ) ) {\r
-- errorText_ = "RtApiWasapi::getDeviceInfo: Unable to open default device property store.";\r
-- goto Exit;\r
-- }\r
-- PropVariantInit( &defaultDeviceNameProp );\r
--\r
-- hr = defaultDevicePropStore->GetValue( PKEY_Device_FriendlyName, &defaultDeviceNameProp );\r
-- if ( FAILED( hr ) ) {\r
-- errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve default device property: PKEY_Device_FriendlyName.";\r
-- goto Exit;\r
-- }\r
--\r
-- defaultDeviceName = convertCharPointerToStdString(defaultDeviceNameProp.pwszVal);\r
--\r
-- // name\r
-- hr = devicePtr->OpenPropertyStore( STGM_READ, &devicePropStore );\r
-- if ( FAILED( hr ) ) {\r
-- errorText_ = "RtApiWasapi::getDeviceInfo: Unable to open device property store.";\r
-- goto Exit;\r
-- }\r
--\r
-- PropVariantInit( &deviceNameProp );\r
--\r
-- hr = devicePropStore->GetValue( PKEY_Device_FriendlyName, &deviceNameProp );\r
-- if ( FAILED( hr ) ) {\r
-- errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve device property: PKEY_Device_FriendlyName.";\r
-- goto Exit;\r
-- }\r
--\r
-- info.name =convertCharPointerToStdString(deviceNameProp.pwszVal);\r
--\r
-- // is default\r
-- if ( isCaptureDevice ) {\r
-- info.isDefaultInput = info.name == defaultDeviceName;\r
-- info.isDefaultOutput = false;\r
-- }\r
-- else {\r
-- info.isDefaultInput = false;\r
-- info.isDefaultOutput = info.name == defaultDeviceName;\r
-- }\r
--\r
-- // channel count\r
-- hr = devicePtr->Activate( __uuidof( IAudioClient ), CLSCTX_ALL, NULL, ( void** ) &audioClient );\r
-- if ( FAILED( hr ) ) {\r
-- errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve device audio client.";\r
-- goto Exit;\r
-- }\r
--\r
-- hr = audioClient->GetMixFormat( &deviceFormat );\r
-- if ( FAILED( hr ) ) {\r
-- errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve device mix format.";\r
-- goto Exit;\r
-- }\r
--\r
-- if ( isCaptureDevice ) {\r
-- info.inputChannels = deviceFormat->nChannels;\r
-- info.outputChannels = 0;\r
-- info.duplexChannels = 0;\r
-- }\r
-- else {\r
-- info.inputChannels = 0;\r
-- info.outputChannels = deviceFormat->nChannels;\r
-- info.duplexChannels = 0;\r
-- }\r
--\r
-- // sample rates\r
-- info.sampleRates.clear();\r
--\r
-- // allow support for all sample rates as we have a built-in sample rate converter\r
-- for ( unsigned int i = 0; i < MAX_SAMPLE_RATES; i++ ) {\r
-- info.sampleRates.push_back( SAMPLE_RATES[i] );\r
-- }\r
-- info.preferredSampleRate = deviceFormat->nSamplesPerSec;\r
--\r
-- // native format\r
-- info.nativeFormats = 0;\r
--\r
-- if ( deviceFormat->wFormatTag == WAVE_FORMAT_IEEE_FLOAT ||\r
-- ( deviceFormat->wFormatTag == WAVE_FORMAT_EXTENSIBLE &&\r
-- ( ( WAVEFORMATEXTENSIBLE* ) deviceFormat )->SubFormat == KSDATAFORMAT_SUBTYPE_IEEE_FLOAT ) )\r
-- {\r
-- if ( deviceFormat->wBitsPerSample == 32 ) {\r
-- info.nativeFormats |= RTAUDIO_FLOAT32;\r
-- }\r
-- else if ( deviceFormat->wBitsPerSample == 64 ) {\r
-- info.nativeFormats |= RTAUDIO_FLOAT64;\r
-- }\r
-- }\r
-- else if ( deviceFormat->wFormatTag == WAVE_FORMAT_PCM ||\r
-- ( deviceFormat->wFormatTag == WAVE_FORMAT_EXTENSIBLE &&\r
-- ( ( WAVEFORMATEXTENSIBLE* ) deviceFormat )->SubFormat == KSDATAFORMAT_SUBTYPE_PCM ) )\r
-- {\r
-- if ( deviceFormat->wBitsPerSample == 8 ) {\r
-- info.nativeFormats |= RTAUDIO_SINT8;\r
-- }\r
-- else if ( deviceFormat->wBitsPerSample == 16 ) {\r
-- info.nativeFormats |= RTAUDIO_SINT16;\r
-- }\r
-- else if ( deviceFormat->wBitsPerSample == 24 ) {\r
-- info.nativeFormats |= RTAUDIO_SINT24;\r
-- }\r
-- else if ( deviceFormat->wBitsPerSample == 32 ) {\r
-- info.nativeFormats |= RTAUDIO_SINT32;\r
-- }\r
-- }\r
--\r
-- // probed\r
-- info.probed = true;\r
--\r
--Exit:\r
-- // release all references\r
-- PropVariantClear( &deviceNameProp );\r
-- PropVariantClear( &defaultDeviceNameProp );\r
--\r
-- SAFE_RELEASE( captureDevices );\r
-- SAFE_RELEASE( renderDevices );\r
-- SAFE_RELEASE( devicePtr );\r
-- SAFE_RELEASE( defaultDevicePtr );\r
-- SAFE_RELEASE( audioClient );\r
-- SAFE_RELEASE( devicePropStore );\r
-- SAFE_RELEASE( defaultDevicePropStore );\r
--\r
-- CoTaskMemFree( deviceFormat );\r
-- CoTaskMemFree( closestMatchFormat );\r
--\r
-- if ( !errorText_.empty() )\r
-- error( errorType );\r
-- return info;\r
--}\r
--\r
--//-----------------------------------------------------------------------------\r
--\r
--unsigned int RtApiWasapi::getDefaultOutputDevice( void )\r
--{\r
-- for ( unsigned int i = 0; i < getDeviceCount(); i++ ) {\r
-- if ( getDeviceInfo( i ).isDefaultOutput ) {\r
-- return i;\r
-- }\r
-- }\r
--\r
-- return 0;\r
--}\r
--\r
--//-----------------------------------------------------------------------------\r
--\r
--unsigned int RtApiWasapi::getDefaultInputDevice( void )\r
--{\r
-- for ( unsigned int i = 0; i < getDeviceCount(); i++ ) {\r
-- if ( getDeviceInfo( i ).isDefaultInput ) {\r
-- return i;\r
-- }\r
-- }\r
--\r
-- return 0;\r
--}\r
--\r
--//-----------------------------------------------------------------------------\r
--\r
--void RtApiWasapi::closeStream( void )\r
--{\r
-- if ( stream_.state == STREAM_CLOSED ) {\r
-- errorText_ = "RtApiWasapi::closeStream: No open stream to close.";\r
-- error( RtAudioError::WARNING );\r
-- return;\r
-- }\r
--\r
-- if ( stream_.state != STREAM_STOPPED )\r
-- stopStream();\r
--\r
-- // clean up stream memory\r
-- SAFE_RELEASE( ( ( WasapiHandle* ) stream_.apiHandle )->captureAudioClient )\r
-- SAFE_RELEASE( ( ( WasapiHandle* ) stream_.apiHandle )->renderAudioClient )\r
--\r
-- SAFE_RELEASE( ( ( WasapiHandle* ) stream_.apiHandle )->captureClient )\r
-- SAFE_RELEASE( ( ( WasapiHandle* ) stream_.apiHandle )->renderClient )\r
--\r
-- if ( ( ( WasapiHandle* ) stream_.apiHandle )->captureEvent )\r
-- CloseHandle( ( ( WasapiHandle* ) stream_.apiHandle )->captureEvent );\r
--\r
-- if ( ( ( WasapiHandle* ) stream_.apiHandle )->renderEvent )\r
-- CloseHandle( ( ( WasapiHandle* ) stream_.apiHandle )->renderEvent );\r
--\r
-- delete ( WasapiHandle* ) stream_.apiHandle;\r
-- stream_.apiHandle = NULL;\r
--\r
-- for ( int i = 0; i < 2; i++ ) {\r
-- if ( stream_.userBuffer[i] ) {\r
-- free( stream_.userBuffer[i] );\r
-- stream_.userBuffer[i] = 0;\r
-- }\r
-- }\r
--\r
-- if ( stream_.deviceBuffer ) {\r
-- free( stream_.deviceBuffer );\r
-- stream_.deviceBuffer = 0;\r
-- }\r
--\r
-- // update stream state\r
-- stream_.state = STREAM_CLOSED;\r
--}\r
--\r
--//-----------------------------------------------------------------------------\r
--\r
--void RtApiWasapi::startStream( void )\r
--{\r
-- verifyStream();\r
--\r
-- if ( stream_.state == STREAM_RUNNING ) {\r
-- errorText_ = "RtApiWasapi::startStream: The stream is already running.";\r
-- error( RtAudioError::WARNING );\r
-- return;\r
-- }\r
--\r
-- // update stream state\r
-- stream_.state = STREAM_RUNNING;\r
--\r
-- // create WASAPI stream thread\r
-- stream_.callbackInfo.thread = ( ThreadHandle ) CreateThread( NULL, 0, runWasapiThread, this, CREATE_SUSPENDED, NULL );\r
--\r
-- if ( !stream_.callbackInfo.thread ) {\r
-- errorText_ = "RtApiWasapi::startStream: Unable to instantiate callback thread.";\r
-- error( RtAudioError::THREAD_ERROR );\r
-- }\r
-- else {\r
-- SetThreadPriority( ( void* ) stream_.callbackInfo.thread, stream_.callbackInfo.priority );\r
-- ResumeThread( ( void* ) stream_.callbackInfo.thread );\r
-- }\r
--}\r
--\r
--//-----------------------------------------------------------------------------\r
--\r
--void RtApiWasapi::stopStream( void )\r
--{\r
-- verifyStream();\r
--\r
-- if ( stream_.state == STREAM_STOPPED ) {\r
-- errorText_ = "RtApiWasapi::stopStream: The stream is already stopped.";\r
-- error( RtAudioError::WARNING );\r
-- return;\r
-- }\r
--\r
-- // inform stream thread by setting stream state to STREAM_STOPPING\r
-- stream_.state = STREAM_STOPPING;\r
--\r
-- // wait until stream thread is stopped\r
-- while( stream_.state != STREAM_STOPPED ) {\r
-- Sleep( 1 );\r
-- }\r
--\r
-- // Wait for the last buffer to play before stopping.\r
-- Sleep( 1000 * stream_.bufferSize / stream_.sampleRate );\r
--\r
-- // stop capture client if applicable\r
-- if ( ( ( WasapiHandle* ) stream_.apiHandle )->captureAudioClient ) {\r
-- HRESULT hr = ( ( WasapiHandle* ) stream_.apiHandle )->captureAudioClient->Stop();\r
-- if ( FAILED( hr ) ) {\r
-- errorText_ = "RtApiWasapi::stopStream: Unable to stop capture stream.";\r
-- error( RtAudioError::DRIVER_ERROR );\r
-- return;\r
-- }\r
-- }\r
--\r
-- // stop render client if applicable\r
-- if ( ( ( WasapiHandle* ) stream_.apiHandle )->renderAudioClient ) {\r
-- HRESULT hr = ( ( WasapiHandle* ) stream_.apiHandle )->renderAudioClient->Stop();\r
-- if ( FAILED( hr ) ) {\r
-- errorText_ = "RtApiWasapi::stopStream: Unable to stop render stream.";\r
-- error( RtAudioError::DRIVER_ERROR );\r
-- return;\r
-- }\r
-- }\r
--\r
-- // close thread handle\r
-- if ( stream_.callbackInfo.thread && !CloseHandle( ( void* ) stream_.callbackInfo.thread ) ) {\r
-- errorText_ = "RtApiWasapi::stopStream: Unable to close callback thread.";\r
-- error( RtAudioError::THREAD_ERROR );\r
-- return;\r
-- }\r
--\r
-- stream_.callbackInfo.thread = (ThreadHandle) NULL;\r
--}\r
--\r
--//-----------------------------------------------------------------------------\r
--\r
--void RtApiWasapi::abortStream( void )\r
--{\r
-- verifyStream();\r
--\r
-- if ( stream_.state == STREAM_STOPPED ) {\r
-- errorText_ = "RtApiWasapi::abortStream: The stream is already stopped.";\r
-- error( RtAudioError::WARNING );\r
-- return;\r
-- }\r
--\r
-- // inform stream thread by setting stream state to STREAM_STOPPING\r
-- stream_.state = STREAM_STOPPING;\r
--\r
-- // wait until stream thread is stopped\r
-- while ( stream_.state != STREAM_STOPPED ) {\r
-- Sleep( 1 );\r
-- }\r
--\r
-- // stop capture client if applicable\r
-- if ( ( ( WasapiHandle* ) stream_.apiHandle )->captureAudioClient ) {\r
-- HRESULT hr = ( ( WasapiHandle* ) stream_.apiHandle )->captureAudioClient->Stop();\r
-- if ( FAILED( hr ) ) {\r
-- errorText_ = "RtApiWasapi::abortStream: Unable to stop capture stream.";\r
-- error( RtAudioError::DRIVER_ERROR );\r
-- return;\r
-- }\r
-- }\r
--\r
-- // stop render client if applicable\r
-- if ( ( ( WasapiHandle* ) stream_.apiHandle )->renderAudioClient ) {\r
-- HRESULT hr = ( ( WasapiHandle* ) stream_.apiHandle )->renderAudioClient->Stop();\r
-- if ( FAILED( hr ) ) {\r
-- errorText_ = "RtApiWasapi::abortStream: Unable to stop render stream.";\r
-- error( RtAudioError::DRIVER_ERROR );\r
-- return;\r
-- }\r
-- }\r
--\r
-- // close thread handle\r
-- if ( stream_.callbackInfo.thread && !CloseHandle( ( void* ) stream_.callbackInfo.thread ) ) {\r
-- errorText_ = "RtApiWasapi::abortStream: Unable to close callback thread.";\r
-- error( RtAudioError::THREAD_ERROR );\r
-- return;\r
-- }\r
--\r
-- stream_.callbackInfo.thread = (ThreadHandle) NULL;\r
--}\r
--\r
--//-----------------------------------------------------------------------------\r
--\r
--bool RtApiWasapi::probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,\r
-- unsigned int firstChannel, unsigned int sampleRate,\r
-- RtAudioFormat format, unsigned int* bufferSize,\r
-- RtAudio::StreamOptions* options )\r
--{\r
-- bool methodResult = FAILURE;\r
-- unsigned int captureDeviceCount = 0;\r
-- unsigned int renderDeviceCount = 0;\r
--\r
-- IMMDeviceCollection* captureDevices = NULL;\r
-- IMMDeviceCollection* renderDevices = NULL;\r
-- IMMDevice* devicePtr = NULL;\r
-- WAVEFORMATEX* deviceFormat = NULL;\r
-- unsigned int bufferBytes;\r
-- stream_.state = STREAM_STOPPED;\r
--\r
-- // create API Handle if not already created\r
-- if ( !stream_.apiHandle )\r
-- stream_.apiHandle = ( void* ) new WasapiHandle();\r
--\r
-- // Count capture devices\r
-- errorText_.clear();\r
-- RtAudioError::Type errorType = RtAudioError::DRIVER_ERROR;\r
-- HRESULT hr = deviceEnumerator_->EnumAudioEndpoints( eCapture, DEVICE_STATE_ACTIVE, &captureDevices );\r
-- if ( FAILED( hr ) ) {\r
-- errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve capture device collection.";\r
-- goto Exit;\r
-- }\r
--\r
-- hr = captureDevices->GetCount( &captureDeviceCount );\r
-- if ( FAILED( hr ) ) {\r
-- errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve capture device count.";\r
-- goto Exit;\r
-- }\r
--\r
-- // Count render devices\r
-- hr = deviceEnumerator_->EnumAudioEndpoints( eRender, DEVICE_STATE_ACTIVE, &renderDevices );\r
-- if ( FAILED( hr ) ) {\r
-- errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve render device collection.";\r
-- goto Exit;\r
-- }\r
--\r
-- hr = renderDevices->GetCount( &renderDeviceCount );\r
-- if ( FAILED( hr ) ) {\r
-- errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve render device count.";\r
-- goto Exit;\r
-- }\r
--\r
-- // validate device index\r
-- if ( device >= captureDeviceCount + renderDeviceCount ) {\r
-- errorType = RtAudioError::INVALID_USE;\r
-- errorText_ = "RtApiWasapi::probeDeviceOpen: Invalid device index.";\r
-- goto Exit;\r
-- }\r
--\r
-- // determine whether index falls within capture or render devices\r
-- if ( device >= renderDeviceCount ) {\r
-- if ( mode != INPUT ) {\r
-- errorType = RtAudioError::INVALID_USE;\r
-- errorText_ = "RtApiWasapi::probeDeviceOpen: Capture device selected as output device.";\r
-- goto Exit;\r
-- }\r
--\r
-- // retrieve captureAudioClient from devicePtr\r
-- IAudioClient*& captureAudioClient = ( ( WasapiHandle* ) stream_.apiHandle )->captureAudioClient;\r
--\r
-- hr = captureDevices->Item( device - renderDeviceCount, &devicePtr );\r
-- if ( FAILED( hr ) ) {\r
-- errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve capture device handle.";\r
-- goto Exit;\r
-- }\r
--\r
-- hr = devicePtr->Activate( __uuidof( IAudioClient ), CLSCTX_ALL,\r
-- NULL, ( void** ) &captureAudioClient );\r
-- if ( FAILED( hr ) ) {\r
-- errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve device audio client.";\r
-- goto Exit;\r
-- }\r
--\r
-- hr = captureAudioClient->GetMixFormat( &deviceFormat );\r
-- if ( FAILED( hr ) ) {\r
-- errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve device mix format.";\r
-- goto Exit;\r
-- }\r
--\r
-- stream_.nDeviceChannels[mode] = deviceFormat->nChannels;\r
-- captureAudioClient->GetStreamLatency( ( long long* ) &stream_.latency[mode] );\r
-- }\r
-- else {\r
-- if ( mode != OUTPUT ) {\r
-- errorType = RtAudioError::INVALID_USE;\r
-- errorText_ = "RtApiWasapi::probeDeviceOpen: Render device selected as input device.";\r
-- goto Exit;\r
-- }\r
--\r
-- // retrieve renderAudioClient from devicePtr\r
-- IAudioClient*& renderAudioClient = ( ( WasapiHandle* ) stream_.apiHandle )->renderAudioClient;\r
--\r
-- hr = renderDevices->Item( device, &devicePtr );\r
-- if ( FAILED( hr ) ) {\r
-- errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve render device handle.";\r
-- goto Exit;\r
-- }\r
--\r
-- hr = devicePtr->Activate( __uuidof( IAudioClient ), CLSCTX_ALL,\r
-- NULL, ( void** ) &renderAudioClient );\r
-- if ( FAILED( hr ) ) {\r
-- errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve device audio client.";\r
-- goto Exit;\r
-- }\r
--\r
-- hr = renderAudioClient->GetMixFormat( &deviceFormat );\r
-- if ( FAILED( hr ) ) {\r
-- errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve device mix format.";\r
-- goto Exit;\r
-- }\r
--\r
-- stream_.nDeviceChannels[mode] = deviceFormat->nChannels;\r
-- renderAudioClient->GetStreamLatency( ( long long* ) &stream_.latency[mode] );\r
-- }\r
--\r
-- // fill stream data\r
-- if ( ( stream_.mode == OUTPUT && mode == INPUT ) ||\r
-- ( stream_.mode == INPUT && mode == OUTPUT ) ) {\r
-- stream_.mode = DUPLEX;\r
-- }\r
-- else {\r
-- stream_.mode = mode;\r
-- }\r
--\r
-- stream_.device[mode] = device;\r
-- stream_.doByteSwap[mode] = false;\r
-- stream_.sampleRate = sampleRate;\r
-- stream_.bufferSize = *bufferSize;\r
-- stream_.nBuffers = 1;\r
-- stream_.nUserChannels[mode] = channels;\r
-- stream_.channelOffset[mode] = firstChannel;\r
-- stream_.userFormat = format;\r
-- stream_.deviceFormat[mode] = getDeviceInfo( device ).nativeFormats;\r
--\r
-- if ( options && options->flags & RTAUDIO_NONINTERLEAVED )\r
-- stream_.userInterleaved = false;\r
-- else\r
-- stream_.userInterleaved = true;\r
-- stream_.deviceInterleaved[mode] = true;\r
--\r
-- // Set flags for buffer conversion.\r
-- stream_.doConvertBuffer[mode] = false;\r
-- if ( stream_.userFormat != stream_.deviceFormat[mode] ||\r
-- stream_.nUserChannels != stream_.nDeviceChannels )\r
-- stream_.doConvertBuffer[mode] = true;\r
-- else if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] &&\r
-- stream_.nUserChannels[mode] > 1 )\r
-- stream_.doConvertBuffer[mode] = true;\r
--\r
-- if ( stream_.doConvertBuffer[mode] )\r
-- setConvertInfo( mode, 0 );\r
--\r
-- // Allocate necessary internal buffers\r
-- bufferBytes = stream_.nUserChannels[mode] * stream_.bufferSize * formatBytes( stream_.userFormat );\r
--\r
-- stream_.userBuffer[mode] = ( char* ) calloc( bufferBytes, 1 );\r
-- if ( !stream_.userBuffer[mode] ) {\r
-- errorType = RtAudioError::MEMORY_ERROR;\r
-- errorText_ = "RtApiWasapi::probeDeviceOpen: Error allocating user buffer memory.";\r
-- goto Exit;\r
-- }\r
--\r
-- if ( options && options->flags & RTAUDIO_SCHEDULE_REALTIME )\r
-- stream_.callbackInfo.priority = 15;\r
-- else\r
-- stream_.callbackInfo.priority = 0;\r
--\r
-- ///! TODO: RTAUDIO_MINIMIZE_LATENCY // Provide stream buffers directly to callback\r
-- ///! TODO: RTAUDIO_HOG_DEVICE // Exclusive mode\r
--\r
-- methodResult = SUCCESS;\r
--\r
--Exit:\r
-- //clean up\r
-- SAFE_RELEASE( captureDevices );\r
-- SAFE_RELEASE( renderDevices );\r
-- SAFE_RELEASE( devicePtr );\r
-- CoTaskMemFree( deviceFormat );\r
--\r
-- // if method failed, close the stream\r
-- if ( methodResult == FAILURE )\r
-- closeStream();\r
--\r
-- if ( !errorText_.empty() )\r
-- error( errorType );\r
-- return methodResult;\r
--}\r
--\r
--//=============================================================================\r
--\r
--DWORD WINAPI RtApiWasapi::runWasapiThread( void* wasapiPtr )\r
--{\r
-- if ( wasapiPtr )\r
-- ( ( RtApiWasapi* ) wasapiPtr )->wasapiThread();\r
--\r
-- return 0;\r
--}\r
--\r
--DWORD WINAPI RtApiWasapi::stopWasapiThread( void* wasapiPtr )\r
--{\r
-- if ( wasapiPtr )\r
-- ( ( RtApiWasapi* ) wasapiPtr )->stopStream();\r
--\r
-- return 0;\r
--}\r
--\r
--DWORD WINAPI RtApiWasapi::abortWasapiThread( void* wasapiPtr )\r
--{\r
-- if ( wasapiPtr )\r
-- ( ( RtApiWasapi* ) wasapiPtr )->abortStream();\r
--\r
-- return 0;\r
--}\r
--\r
--//-----------------------------------------------------------------------------\r
--\r
--void RtApiWasapi::wasapiThread()\r
--{\r
-- // as this is a new thread, we must CoInitialize it\r
-- CoInitialize( NULL );\r
--\r
-- HRESULT hr;\r
--\r
-- IAudioClient* captureAudioClient = ( ( WasapiHandle* ) stream_.apiHandle )->captureAudioClient;\r
-- IAudioClient* renderAudioClient = ( ( WasapiHandle* ) stream_.apiHandle )->renderAudioClient;\r
-- IAudioCaptureClient* captureClient = ( ( WasapiHandle* ) stream_.apiHandle )->captureClient;\r
-- IAudioRenderClient* renderClient = ( ( WasapiHandle* ) stream_.apiHandle )->renderClient;\r
-- HANDLE captureEvent = ( ( WasapiHandle* ) stream_.apiHandle )->captureEvent;\r
-- HANDLE renderEvent = ( ( WasapiHandle* ) stream_.apiHandle )->renderEvent;\r
--\r
-- WAVEFORMATEX* captureFormat = NULL;\r
-- WAVEFORMATEX* renderFormat = NULL;\r
-- float captureSrRatio = 0.0f;\r
-- float renderSrRatio = 0.0f;\r
-- WasapiBuffer captureBuffer;\r
-- WasapiBuffer renderBuffer;\r
--\r
-- // declare local stream variables\r
-- RtAudioCallback callback = ( RtAudioCallback ) stream_.callbackInfo.callback;\r
-- BYTE* streamBuffer = NULL;\r
-- unsigned long captureFlags = 0;\r
-- unsigned int bufferFrameCount = 0;\r
-- unsigned int numFramesPadding = 0;\r
-- unsigned int convBufferSize = 0;\r
-- bool callbackPushed = false;\r
-- bool callbackPulled = false;\r
-- bool callbackStopped = false;\r
-- int callbackResult = 0;\r
--\r
-- // convBuffer is used to store converted buffers between WASAPI and the user\r
-- char* convBuffer = NULL;\r
-- unsigned int convBuffSize = 0;\r
-- unsigned int deviceBuffSize = 0;\r
--\r
-- errorText_.clear();\r
-- RtAudioError::Type errorType = RtAudioError::DRIVER_ERROR;\r
--\r
-- // Attempt to assign "Pro Audio" characteristic to thread\r
-- HMODULE AvrtDll = LoadLibrary( (LPCTSTR) "AVRT.dll" );\r
-- if ( AvrtDll ) {\r
-- DWORD taskIndex = 0;\r
-- TAvSetMmThreadCharacteristicsPtr AvSetMmThreadCharacteristicsPtr = ( TAvSetMmThreadCharacteristicsPtr ) GetProcAddress( AvrtDll, "AvSetMmThreadCharacteristicsW" );\r
-- AvSetMmThreadCharacteristicsPtr( L"Pro Audio", &taskIndex );\r
-- FreeLibrary( AvrtDll );\r
-- }\r
--\r
-- // start capture stream if applicable\r
-- if ( captureAudioClient ) {\r
-- hr = captureAudioClient->GetMixFormat( &captureFormat );\r
-- if ( FAILED( hr ) ) {\r
-- errorText_ = "RtApiWasapi::wasapiThread: Unable to retrieve device mix format.";\r
-- goto Exit;\r
-- }\r
--\r
-- captureSrRatio = ( ( float ) captureFormat->nSamplesPerSec / stream_.sampleRate );\r
--\r
-- // initialize capture stream according to desire buffer size\r
-- float desiredBufferSize = stream_.bufferSize * captureSrRatio;\r
-- REFERENCE_TIME desiredBufferPeriod = ( REFERENCE_TIME ) ( ( float ) desiredBufferSize * 10000000 / captureFormat->nSamplesPerSec );\r
--\r
-- if ( !captureClient ) {\r
-- hr = captureAudioClient->Initialize( AUDCLNT_SHAREMODE_SHARED,\r
-- AUDCLNT_STREAMFLAGS_EVENTCALLBACK,\r
-- desiredBufferPeriod,\r
-- desiredBufferPeriod,\r
-- captureFormat,\r
-- NULL );\r
-- if ( FAILED( hr ) ) {\r
-- errorText_ = "RtApiWasapi::wasapiThread: Unable to initialize capture audio client.";\r
-- goto Exit;\r
-- }\r
--\r
-- hr = captureAudioClient->GetService( __uuidof( IAudioCaptureClient ),\r
-- ( void** ) &captureClient );\r
-- if ( FAILED( hr ) ) {\r
-- errorText_ = "RtApiWasapi::wasapiThread: Unable to retrieve capture client handle.";\r
-- goto Exit;\r
-- }\r
--\r
-- // configure captureEvent to trigger on every available capture buffer\r
-- captureEvent = CreateEvent( NULL, FALSE, FALSE, NULL );\r
-- if ( !captureEvent ) {\r
-- errorType = RtAudioError::SYSTEM_ERROR;\r
-- errorText_ = "RtApiWasapi::wasapiThread: Unable to create capture event.";\r
-- goto Exit;\r
-- }\r
--\r
-- hr = captureAudioClient->SetEventHandle( captureEvent );\r
-- if ( FAILED( hr ) ) {\r
-- errorText_ = "RtApiWasapi::wasapiThread: Unable to set capture event handle.";\r
-- goto Exit;\r
-- }\r
--\r
-- ( ( WasapiHandle* ) stream_.apiHandle )->captureClient = captureClient;\r
-- ( ( WasapiHandle* ) stream_.apiHandle )->captureEvent = captureEvent;\r
-- }\r
--\r
-- unsigned int inBufferSize = 0;\r
-- hr = captureAudioClient->GetBufferSize( &inBufferSize );\r
-- if ( FAILED( hr ) ) {\r
-- errorText_ = "RtApiWasapi::wasapiThread: Unable to get capture buffer size.";\r
-- goto Exit;\r
-- }\r
--\r
-- // scale outBufferSize according to stream->user sample rate ratio\r
-- unsigned int outBufferSize = ( unsigned int ) ( stream_.bufferSize * captureSrRatio ) * stream_.nDeviceChannels[INPUT];\r
-- inBufferSize *= stream_.nDeviceChannels[INPUT];\r
--\r
-- // set captureBuffer size\r
-- captureBuffer.setBufferSize( inBufferSize + outBufferSize, formatBytes( stream_.deviceFormat[INPUT] ) );\r
--\r
-- // reset the capture stream\r
-- hr = captureAudioClient->Reset();\r
-- if ( FAILED( hr ) ) {\r
-- errorText_ = "RtApiWasapi::wasapiThread: Unable to reset capture stream.";\r
-- goto Exit;\r
-- }\r
--\r
-- // start the capture stream\r
-- hr = captureAudioClient->Start();\r
-- if ( FAILED( hr ) ) {\r
-- errorText_ = "RtApiWasapi::wasapiThread: Unable to start capture stream.";\r
-- goto Exit;\r
-- }\r
-- }\r
--\r
-- // start render stream if applicable\r
-- if ( renderAudioClient ) {\r
-- hr = renderAudioClient->GetMixFormat( &renderFormat );\r
-- if ( FAILED( hr ) ) {\r
-- errorText_ = "RtApiWasapi::wasapiThread: Unable to retrieve device mix format.";\r
-- goto Exit;\r
-- }\r
--\r
-- renderSrRatio = ( ( float ) renderFormat->nSamplesPerSec / stream_.sampleRate );\r
--\r
-- // initialize render stream according to desire buffer size\r
-- float desiredBufferSize = stream_.bufferSize * renderSrRatio;\r
-- REFERENCE_TIME desiredBufferPeriod = ( REFERENCE_TIME ) ( ( float ) desiredBufferSize * 10000000 / renderFormat->nSamplesPerSec );\r
--\r
-- if ( !renderClient ) {\r
-- hr = renderAudioClient->Initialize( AUDCLNT_SHAREMODE_SHARED,\r
-- AUDCLNT_STREAMFLAGS_EVENTCALLBACK,\r
-- desiredBufferPeriod,\r
-- desiredBufferPeriod,\r
-- renderFormat,\r
-- NULL );\r
-- if ( FAILED( hr ) ) {\r
-- errorText_ = "RtApiWasapi::wasapiThread: Unable to initialize render audio client.";\r
-- goto Exit;\r
-- }\r
--\r
-- hr = renderAudioClient->GetService( __uuidof( IAudioRenderClient ),\r
-- ( void** ) &renderClient );\r
-- if ( FAILED( hr ) ) {\r
-- errorText_ = "RtApiWasapi::wasapiThread: Unable to retrieve render client handle.";\r
-- goto Exit;\r
-- }\r
--\r
-- // configure renderEvent to trigger on every available render buffer\r
-- renderEvent = CreateEvent( NULL, FALSE, FALSE, NULL );\r
-- if ( !renderEvent ) {\r
-- errorType = RtAudioError::SYSTEM_ERROR;\r
-- errorText_ = "RtApiWasapi::wasapiThread: Unable to create render event.";\r
-- goto Exit;\r
-- }\r
--\r
-- hr = renderAudioClient->SetEventHandle( renderEvent );\r
-- if ( FAILED( hr ) ) {\r
-- errorText_ = "RtApiWasapi::wasapiThread: Unable to set render event handle.";\r
-- goto Exit;\r
-- }\r
--\r
-- ( ( WasapiHandle* ) stream_.apiHandle )->renderClient = renderClient;\r
-- ( ( WasapiHandle* ) stream_.apiHandle )->renderEvent = renderEvent;\r
-- }\r
--\r
-- unsigned int outBufferSize = 0;\r
-- hr = renderAudioClient->GetBufferSize( &outBufferSize );\r
-- if ( FAILED( hr ) ) {\r
-- errorText_ = "RtApiWasapi::wasapiThread: Unable to get render buffer size.";\r
-- goto Exit;\r
-- }\r
--\r
-- // scale inBufferSize according to user->stream sample rate ratio\r
-- unsigned int inBufferSize = ( unsigned int ) ( stream_.bufferSize * renderSrRatio ) * stream_.nDeviceChannels[OUTPUT];\r
-- outBufferSize *= stream_.nDeviceChannels[OUTPUT];\r
--\r
-- // set renderBuffer size\r
-- renderBuffer.setBufferSize( inBufferSize + outBufferSize, formatBytes( stream_.deviceFormat[OUTPUT] ) );\r
--\r
-- // reset the render stream\r
-- hr = renderAudioClient->Reset();\r
-- if ( FAILED( hr ) ) {\r
-- errorText_ = "RtApiWasapi::wasapiThread: Unable to reset render stream.";\r
-- goto Exit;\r
-- }\r
--\r
-- // start the render stream\r
-- hr = renderAudioClient->Start();\r
-- if ( FAILED( hr ) ) {\r
-- errorText_ = "RtApiWasapi::wasapiThread: Unable to start render stream.";\r
-- goto Exit;\r
-- }\r
-- }\r
--\r
-- if ( stream_.mode == INPUT ) {\r
-- convBuffSize = ( size_t ) ( stream_.bufferSize * captureSrRatio ) * stream_.nDeviceChannels[INPUT] * formatBytes( stream_.deviceFormat[INPUT] );\r
-- deviceBuffSize = stream_.bufferSize * stream_.nDeviceChannels[INPUT] * formatBytes( stream_.deviceFormat[INPUT] );\r
-- }\r
-- else if ( stream_.mode == OUTPUT ) {\r
-- convBuffSize = ( size_t ) ( stream_.bufferSize * renderSrRatio ) * stream_.nDeviceChannels[OUTPUT] * formatBytes( stream_.deviceFormat[OUTPUT] );\r
-- deviceBuffSize = stream_.bufferSize * stream_.nDeviceChannels[OUTPUT] * formatBytes( stream_.deviceFormat[OUTPUT] );\r
-- }\r
-- else if ( stream_.mode == DUPLEX ) {\r
-- convBuffSize = std::max( ( size_t ) ( stream_.bufferSize * captureSrRatio ) * stream_.nDeviceChannels[INPUT] * formatBytes( stream_.deviceFormat[INPUT] ),\r
-- ( size_t ) ( stream_.bufferSize * renderSrRatio ) * stream_.nDeviceChannels[OUTPUT] * formatBytes( stream_.deviceFormat[OUTPUT] ) );\r
-- deviceBuffSize = std::max( stream_.bufferSize * stream_.nDeviceChannels[INPUT] * formatBytes( stream_.deviceFormat[INPUT] ),\r
-- stream_.bufferSize * stream_.nDeviceChannels[OUTPUT] * formatBytes( stream_.deviceFormat[OUTPUT] ) );\r
-- }\r
--\r
-- convBuffer = ( char* ) malloc( convBuffSize );\r
-- stream_.deviceBuffer = ( char* ) malloc( deviceBuffSize );\r
-- if ( !convBuffer || !stream_.deviceBuffer ) {\r
-- errorType = RtAudioError::MEMORY_ERROR;\r
-- errorText_ = "RtApiWasapi::wasapiThread: Error allocating device buffer memory.";\r
-- goto Exit;\r
-- }\r
--\r
-- // stream process loop\r
-- while ( stream_.state != STREAM_STOPPING ) {\r
-- if ( !callbackPulled ) {\r
-- // Callback Input\r
-- // ==============\r
-- // 1. Pull callback buffer from inputBuffer\r
-- // 2. If 1. was successful: Convert callback buffer to user sample rate and channel count\r
-- // Convert callback buffer to user format\r
--\r
-- if ( captureAudioClient ) {\r
-- // Pull callback buffer from inputBuffer\r
-- callbackPulled = captureBuffer.pullBuffer( convBuffer,\r
-- ( unsigned int ) ( stream_.bufferSize * captureSrRatio ) * stream_.nDeviceChannels[INPUT],\r
-- stream_.deviceFormat[INPUT] );\r
--\r
-- if ( callbackPulled ) {\r
-- // Convert callback buffer to user sample rate\r
-- convertBufferWasapi( stream_.deviceBuffer,\r
-- convBuffer,\r
-- stream_.nDeviceChannels[INPUT],\r
-- captureFormat->nSamplesPerSec,\r
-- stream_.sampleRate,\r
-- ( unsigned int ) ( stream_.bufferSize * captureSrRatio ),\r
-- convBufferSize,\r
-- stream_.deviceFormat[INPUT] );\r
--\r
-- if ( stream_.doConvertBuffer[INPUT] ) {\r
-- // Convert callback buffer to user format\r
-- convertBuffer( stream_.userBuffer[INPUT],\r
-- stream_.deviceBuffer,\r
-- stream_.convertInfo[INPUT] );\r
-- }\r
-- else {\r
-- // no further conversion, simple copy deviceBuffer to userBuffer\r
-- memcpy( stream_.userBuffer[INPUT],\r
-- stream_.deviceBuffer,\r
-- stream_.bufferSize * stream_.nUserChannels[INPUT] * formatBytes( stream_.userFormat ) );\r
-- }\r
-- }\r
-- }\r
-- else {\r
-- // if there is no capture stream, set callbackPulled flag\r
-- callbackPulled = true;\r
-- }\r
--\r
-- // Execute Callback\r
-- // ================\r
-- // 1. Execute user callback method\r
-- // 2. Handle return value from callback\r
--\r
-- // if callback has not requested the stream to stop\r
-- if ( callbackPulled && !callbackStopped ) {\r
-- // Execute user callback method\r
-- callbackResult = callback( stream_.userBuffer[OUTPUT],\r
-- stream_.userBuffer[INPUT],\r
-- stream_.bufferSize,\r
-- getStreamTime(),\r
-- captureFlags & AUDCLNT_BUFFERFLAGS_DATA_DISCONTINUITY ? RTAUDIO_INPUT_OVERFLOW : 0,\r
-- stream_.callbackInfo.userData );\r
--\r
-- // Handle return value from callback\r
-- if ( callbackResult == 1 ) {\r
-- // instantiate a thread to stop this thread\r
-- HANDLE threadHandle = CreateThread( NULL, 0, stopWasapiThread, this, 0, NULL );\r
-- if ( !threadHandle ) {\r
-- errorType = RtAudioError::THREAD_ERROR;\r
-- errorText_ = "RtApiWasapi::wasapiThread: Unable to instantiate stream stop thread.";\r
-- goto Exit;\r
-- }\r
-- else if ( !CloseHandle( threadHandle ) ) {\r
-- errorType = RtAudioError::THREAD_ERROR;\r
-- errorText_ = "RtApiWasapi::wasapiThread: Unable to close stream stop thread handle.";\r
-- goto Exit;\r
-- }\r
--\r
-- callbackStopped = true;\r
-- }\r
-- else if ( callbackResult == 2 ) {\r
-- // instantiate a thread to stop this thread\r
-- HANDLE threadHandle = CreateThread( NULL, 0, abortWasapiThread, this, 0, NULL );\r
-- if ( !threadHandle ) {\r
-- errorType = RtAudioError::THREAD_ERROR;\r
-- errorText_ = "RtApiWasapi::wasapiThread: Unable to instantiate stream abort thread.";\r
-- goto Exit;\r
-- }\r
-- else if ( !CloseHandle( threadHandle ) ) {\r
-- errorType = RtAudioError::THREAD_ERROR;\r
-- errorText_ = "RtApiWasapi::wasapiThread: Unable to close stream abort thread handle.";\r
-- goto Exit;\r
-- }\r
--\r
-- callbackStopped = true;\r
-- }\r
-- }\r
-- }\r
--\r
-- // Callback Output\r
-- // ===============\r
-- // 1. Convert callback buffer to stream format\r
-- // 2. Convert callback buffer to stream sample rate and channel count\r
-- // 3. Push callback buffer into outputBuffer\r
--\r
-- if ( renderAudioClient && callbackPulled ) {\r
-- if ( stream_.doConvertBuffer[OUTPUT] ) {\r
-- // Convert callback buffer to stream format\r
-- convertBuffer( stream_.deviceBuffer,\r
-- stream_.userBuffer[OUTPUT],\r
-- stream_.convertInfo[OUTPUT] );\r
--\r
-- }\r
--\r
-- // Convert callback buffer to stream sample rate\r
-- convertBufferWasapi( convBuffer,\r
-- stream_.deviceBuffer,\r
-- stream_.nDeviceChannels[OUTPUT],\r
-- stream_.sampleRate,\r
-- renderFormat->nSamplesPerSec,\r
-- stream_.bufferSize,\r
-- convBufferSize,\r
-- stream_.deviceFormat[OUTPUT] );\r
--\r
-- // Push callback buffer into outputBuffer\r
-- callbackPushed = renderBuffer.pushBuffer( convBuffer,\r
-- convBufferSize * stream_.nDeviceChannels[OUTPUT],\r
-- stream_.deviceFormat[OUTPUT] );\r
-- }\r
-- else {\r
-- // if there is no render stream, set callbackPushed flag\r
-- callbackPushed = true;\r
-- }\r
--\r
-- // Stream Capture\r
-- // ==============\r
-- // 1. Get capture buffer from stream\r
-- // 2. Push capture buffer into inputBuffer\r
-- // 3. If 2. was successful: Release capture buffer\r
--\r
-- if ( captureAudioClient ) {\r
-- // if the callback input buffer was not pulled from captureBuffer, wait for next capture event\r
-- if ( !callbackPulled ) {\r
-- WaitForSingleObject( captureEvent, INFINITE );\r
-- }\r
--\r
-- // Get capture buffer from stream\r
-- hr = captureClient->GetBuffer( &streamBuffer,\r
-- &bufferFrameCount,\r
-- &captureFlags, NULL, NULL );\r
-- if ( FAILED( hr ) ) {\r
-- errorText_ = "RtApiWasapi::wasapiThread: Unable to retrieve capture buffer.";\r
-- goto Exit;\r
-- }\r
--\r
-- if ( bufferFrameCount != 0 ) {\r
-- // Push capture buffer into inputBuffer\r
-- if ( captureBuffer.pushBuffer( ( char* ) streamBuffer,\r
-- bufferFrameCount * stream_.nDeviceChannels[INPUT],\r
-- stream_.deviceFormat[INPUT] ) )\r
-- {\r
-- // Release capture buffer\r
-- hr = captureClient->ReleaseBuffer( bufferFrameCount );\r
-- if ( FAILED( hr ) ) {\r
-- errorText_ = "RtApiWasapi::wasapiThread: Unable to release capture buffer.";\r
-- goto Exit;\r
-- }\r
-- }\r
-- else\r
-- {\r
-- // Inform WASAPI that capture was unsuccessful\r
-- hr = captureClient->ReleaseBuffer( 0 );\r
-- if ( FAILED( hr ) ) {\r
-- errorText_ = "RtApiWasapi::wasapiThread: Unable to release capture buffer.";\r
-- goto Exit;\r
-- }\r
-- }\r
-- }\r
-- else\r
-- {\r
-- // Inform WASAPI that capture was unsuccessful\r
-- hr = captureClient->ReleaseBuffer( 0 );\r
-- if ( FAILED( hr ) ) {\r
-- errorText_ = "RtApiWasapi::wasapiThread: Unable to release capture buffer.";\r
-- goto Exit;\r
-- }\r
-- }\r
-- }\r
--\r
-- // Stream Render\r
-- // =============\r
-- // 1. Get render buffer from stream\r
-- // 2. Pull next buffer from outputBuffer\r
-- // 3. If 2. was successful: Fill render buffer with next buffer\r
-- // Release render buffer\r
--\r
-- if ( renderAudioClient ) {\r
-- // if the callback output buffer was not pushed to renderBuffer, wait for next render event\r
-- if ( callbackPulled && !callbackPushed ) {\r
-- WaitForSingleObject( renderEvent, INFINITE );\r
-- }\r
--\r
-- // Get render buffer from stream\r
-- hr = renderAudioClient->GetBufferSize( &bufferFrameCount );\r
-- if ( FAILED( hr ) ) {\r
-- errorText_ = "RtApiWasapi::wasapiThread: Unable to retrieve render buffer size.";\r
-- goto Exit;\r
-- }\r
--\r
-- hr = renderAudioClient->GetCurrentPadding( &numFramesPadding );\r
-- if ( FAILED( hr ) ) {\r
-- errorText_ = "RtApiWasapi::wasapiThread: Unable to retrieve render buffer padding.";\r
-- goto Exit;\r
-- }\r
--\r
-- bufferFrameCount -= numFramesPadding;\r
--\r
-- if ( bufferFrameCount != 0 ) {\r
-- hr = renderClient->GetBuffer( bufferFrameCount, &streamBuffer );\r
-- if ( FAILED( hr ) ) {\r
-- errorText_ = "RtApiWasapi::wasapiThread: Unable to retrieve render buffer.";\r
-- goto Exit;\r
-- }\r
--\r
-- // Pull next buffer from outputBuffer\r
-- // Fill render buffer with next buffer\r
-- if ( renderBuffer.pullBuffer( ( char* ) streamBuffer,\r
-- bufferFrameCount * stream_.nDeviceChannels[OUTPUT],\r
-- stream_.deviceFormat[OUTPUT] ) )\r
-- {\r
-- // Release render buffer\r
-- hr = renderClient->ReleaseBuffer( bufferFrameCount, 0 );\r
-- if ( FAILED( hr ) ) {\r
-- errorText_ = "RtApiWasapi::wasapiThread: Unable to release render buffer.";\r
-- goto Exit;\r
-- }\r
-- }\r
-- else\r
-- {\r
-- // Inform WASAPI that render was unsuccessful\r
-- hr = renderClient->ReleaseBuffer( 0, 0 );\r
-- if ( FAILED( hr ) ) {\r
-- errorText_ = "RtApiWasapi::wasapiThread: Unable to release render buffer.";\r
-- goto Exit;\r
-- }\r
-- }\r
-- }\r
-- else\r
-- {\r
-- // Inform WASAPI that render was unsuccessful\r
-- hr = renderClient->ReleaseBuffer( 0, 0 );\r
-- if ( FAILED( hr ) ) {\r
-- errorText_ = "RtApiWasapi::wasapiThread: Unable to release render buffer.";\r
-- goto Exit;\r
-- }\r
-- }\r
-- }\r
--\r
-- // if the callback buffer was pushed renderBuffer reset callbackPulled flag\r
-- if ( callbackPushed ) {\r
-- callbackPulled = false;\r
-- // tick stream time\r
-- RtApi::tickStreamTime();\r
-- }\r
--\r
-- }\r
--\r
--Exit:\r
-- // clean up\r
-- CoTaskMemFree( captureFormat );\r
-- CoTaskMemFree( renderFormat );\r
--\r
-- free ( convBuffer );\r
--\r
-- CoUninitialize();\r
--\r
-- // update stream state\r
-- stream_.state = STREAM_STOPPED;\r
--\r
-- if ( errorText_.empty() )\r
-- return;\r
-- else\r
-- error( errorType );\r
--}\r
--\r
--//******************** End of __WINDOWS_WASAPI__ *********************//\r
--#endif\r
--\r
--\r
--#if defined(__WINDOWS_DS__) // Windows DirectSound API\r
--\r
--// Modified by Robin Davies, October 2005\r
--// - Improvements to DirectX pointer chasing.\r
--// - Bug fix for non-power-of-two Asio granularity used by Edirol PCR-A30.\r
--// - Auto-call CoInitialize for DSOUND and ASIO platforms.\r
--// Various revisions for RtAudio 4.0 by Gary Scavone, April 2007\r
--// Changed device query structure for RtAudio 4.0.7, January 2010\r
--\r
--#include <dsound.h>\r
--#include <assert.h>\r
--#include <algorithm>\r
--\r
--#if defined(__MINGW32__)\r
-- // missing from latest mingw winapi\r
--#define WAVE_FORMAT_96M08 0x00010000 /* 96 kHz, Mono, 8-bit */\r
--#define WAVE_FORMAT_96S08 0x00020000 /* 96 kHz, Stereo, 8-bit */\r
--#define WAVE_FORMAT_96M16 0x00040000 /* 96 kHz, Mono, 16-bit */\r
--#define WAVE_FORMAT_96S16 0x00080000 /* 96 kHz, Stereo, 16-bit */\r
--#endif\r
--\r
--#define MINIMUM_DEVICE_BUFFER_SIZE 32768\r
--\r
--#ifdef _MSC_VER // if Microsoft Visual C++\r
--#pragma comment( lib, "winmm.lib" ) // then, auto-link winmm.lib. Otherwise, it has to be added manually.\r
--#endif\r
--\r
--static inline DWORD dsPointerBetween( DWORD pointer, DWORD laterPointer, DWORD earlierPointer, DWORD bufferSize )\r
--{\r
-- if ( pointer > bufferSize ) pointer -= bufferSize;\r
-- if ( laterPointer < earlierPointer ) laterPointer += bufferSize;\r
-- if ( pointer < earlierPointer ) pointer += bufferSize;\r
-- return pointer >= earlierPointer && pointer < laterPointer;\r
--}\r
--\r
--// A structure to hold various information related to the DirectSound\r
--// API implementation.\r
--struct DsHandle {\r
-- unsigned int drainCounter; // Tracks callback counts when draining\r
-- bool internalDrain; // Indicates if stop is initiated from callback or not.\r
-- void *id[2];\r
-- void *buffer[2];\r
-- bool xrun[2];\r
-- UINT bufferPointer[2];\r
-- DWORD dsBufferSize[2];\r
-- DWORD dsPointerLeadTime[2]; // the number of bytes ahead of the safe pointer to lead by.\r
-- HANDLE condition;\r
--\r
-- DsHandle()\r
-- :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; }\r
--};\r
--\r
--// Declarations for utility functions, callbacks, and structures\r
--// specific to the DirectSound implementation.\r
--static BOOL CALLBACK deviceQueryCallback( LPGUID lpguid,\r
-- LPCTSTR description,\r
-- LPCTSTR module,\r
-- LPVOID lpContext );\r
--\r
--static const char* getErrorString( int code );\r
--\r
--static unsigned __stdcall callbackHandler( void *ptr );\r
--\r
--struct DsDevice {\r
-- LPGUID id[2];\r
-- bool validId[2];\r
-- bool found;\r
-- std::string name;\r
--\r
-- DsDevice()\r
-- : found(false) { validId[0] = false; validId[1] = false; }\r
--};\r
--\r
--struct DsProbeData {\r
-- bool isInput;\r
-- std::vector<struct DsDevice>* dsDevices;\r
--};\r
--\r
--RtApiDs :: RtApiDs()\r
--{\r
-- // Dsound will run both-threaded. If CoInitialize fails, then just\r
-- // accept whatever the mainline chose for a threading model.\r
-- coInitialized_ = false;\r
-- HRESULT hr = CoInitialize( NULL );\r
-- if ( !FAILED( hr ) ) coInitialized_ = true;\r
--}\r
--\r
--RtApiDs :: ~RtApiDs()\r
--{\r
-- if ( coInitialized_ ) CoUninitialize(); // balanced call.\r
-- if ( stream_.state != STREAM_CLOSED ) closeStream();\r
--}\r
--\r
--// The DirectSound default output is always the first device.\r
--unsigned int RtApiDs :: getDefaultOutputDevice( void )\r
--{\r
-- return 0;\r
--}\r
--\r
--// The DirectSound default input is always the first input device,\r
--// which is the first capture device enumerated.\r
--unsigned int RtApiDs :: getDefaultInputDevice( void )\r
--{\r
-- return 0;\r
--}\r
--\r
--unsigned int RtApiDs :: getDeviceCount( void )\r
--{\r
-- // Set query flag for previously found devices to false, so that we\r
-- // can check for any devices that have disappeared.\r
-- for ( unsigned int i=0; i<dsDevices.size(); i++ )\r
-- dsDevices[i].found = false;\r
--\r
-- // Query DirectSound devices.\r
-- struct DsProbeData probeInfo;\r
-- probeInfo.isInput = false;\r
-- probeInfo.dsDevices = &dsDevices;\r
-- HRESULT result = DirectSoundEnumerate( (LPDSENUMCALLBACK) deviceQueryCallback, &probeInfo );\r
-- if ( FAILED( result ) ) {\r
-- errorStream_ << "RtApiDs::getDeviceCount: error (" << getErrorString( result ) << ") enumerating output devices!";\r
-- errorText_ = errorStream_.str();\r
-- error( RtAudioError::WARNING );\r
-- }\r
--\r
-- // Query DirectSoundCapture devices.\r
-- probeInfo.isInput = true;\r
-- result = DirectSoundCaptureEnumerate( (LPDSENUMCALLBACK) deviceQueryCallback, &probeInfo );\r
-- if ( FAILED( result ) ) {\r
-- errorStream_ << "RtApiDs::getDeviceCount: error (" << getErrorString( result ) << ") enumerating input devices!";\r
-- errorText_ = errorStream_.str();\r
-- error( RtAudioError::WARNING );\r
-- }\r
--\r
-- // Clean out any devices that may have disappeared (code update submitted by Eli Zehngut).\r
-- for ( unsigned int i=0; i<dsDevices.size(); ) {\r
-- if ( dsDevices[i].found == false ) dsDevices.erase( dsDevices.begin() + i );\r
-- else i++;\r
-- }\r
--\r
-- return static_cast<unsigned int>(dsDevices.size());\r
--}\r
--\r
--RtAudio::DeviceInfo RtApiDs :: getDeviceInfo( unsigned int device )\r
--{\r
-- RtAudio::DeviceInfo info;\r
-- info.probed = false;\r
--\r
-- if ( dsDevices.size() == 0 ) {\r
-- // Force a query of all devices\r
-- getDeviceCount();\r
-- if ( dsDevices.size() == 0 ) {\r
-- errorText_ = "RtApiDs::getDeviceInfo: no devices found!";\r
-- error( RtAudioError::INVALID_USE );\r
-- return info;\r
-- }\r
-- }\r
--\r
-- if ( device >= dsDevices.size() ) {\r
-- errorText_ = "RtApiDs::getDeviceInfo: device ID is invalid!";\r
-- error( RtAudioError::INVALID_USE );\r
-- return info;\r
-- }\r
--\r
-- HRESULT result;\r
-- if ( dsDevices[ device ].validId[0] == false ) goto probeInput;\r
--\r
-- LPDIRECTSOUND output;\r
-- DSCAPS outCaps;\r
-- result = DirectSoundCreate( dsDevices[ device ].id[0], &output, NULL );\r
-- if ( FAILED( result ) ) {\r
-- errorStream_ << "RtApiDs::getDeviceInfo: error (" << getErrorString( result ) << ") opening output device (" << dsDevices[ device ].name << ")!";\r
-- errorText_ = errorStream_.str();\r
-- error( RtAudioError::WARNING );\r
-- goto probeInput;\r
-- }\r
--\r
-- outCaps.dwSize = sizeof( outCaps );\r
-- result = output->GetCaps( &outCaps );\r
-- if ( FAILED( result ) ) {\r
-- output->Release();\r
-- errorStream_ << "RtApiDs::getDeviceInfo: error (" << getErrorString( result ) << ") getting capabilities!";\r
-- errorText_ = errorStream_.str();\r
-- error( RtAudioError::WARNING );\r
-- goto probeInput;\r
-- }\r
--\r
-- // Get output channel information.\r
-- info.outputChannels = ( outCaps.dwFlags & DSCAPS_PRIMARYSTEREO ) ? 2 : 1;\r
--\r
-- // Get sample rate information.\r
-- info.sampleRates.clear();\r
-- for ( unsigned int k=0; k<MAX_SAMPLE_RATES; k++ ) {\r
-- if ( SAMPLE_RATES[k] >= (unsigned int) outCaps.dwMinSecondarySampleRate &&\r
-- SAMPLE_RATES[k] <= (unsigned int) outCaps.dwMaxSecondarySampleRate ) {\r
-- info.sampleRates.push_back( SAMPLE_RATES[k] );\r
--\r
-- if ( !info.preferredSampleRate || ( SAMPLE_RATES[k] <= 48000 && SAMPLE_RATES[k] > info.preferredSampleRate ) )\r
-- info.preferredSampleRate = SAMPLE_RATES[k];\r
-- }\r
-- }\r
--\r
-- // Get format information.\r
-- if ( outCaps.dwFlags & DSCAPS_PRIMARY16BIT ) info.nativeFormats |= RTAUDIO_SINT16;\r
-- if ( outCaps.dwFlags & DSCAPS_PRIMARY8BIT ) info.nativeFormats |= RTAUDIO_SINT8;\r
--\r
-- output->Release();\r
--\r
-- if ( getDefaultOutputDevice() == device )\r
-- info.isDefaultOutput = true;\r
--\r
-- if ( dsDevices[ device ].validId[1] == false ) {\r
-- info.name = dsDevices[ device ].name;\r
-- info.probed = true;\r
-- return info;\r
-- }\r
--\r
-- probeInput:\r
--\r
-- LPDIRECTSOUNDCAPTURE input;\r
-- result = DirectSoundCaptureCreate( dsDevices[ device ].id[1], &input, NULL );\r
-- if ( FAILED( result ) ) {\r
-- errorStream_ << "RtApiDs::getDeviceInfo: error (" << getErrorString( result ) << ") opening input device (" << dsDevices[ device ].name << ")!";\r
-- errorText_ = errorStream_.str();\r
-- error( RtAudioError::WARNING );\r
-- return info;\r
-- }\r
--\r
-- DSCCAPS inCaps;\r
-- inCaps.dwSize = sizeof( inCaps );\r
-- result = input->GetCaps( &inCaps );\r
-- if ( FAILED( result ) ) {\r
-- input->Release();\r
-- errorStream_ << "RtApiDs::getDeviceInfo: error (" << getErrorString( result ) << ") getting object capabilities (" << dsDevices[ device ].name << ")!";\r
-- errorText_ = errorStream_.str();\r
-- error( RtAudioError::WARNING );\r
-- return info;\r
-- }\r
--\r
-- // Get input channel information.\r
-- info.inputChannels = inCaps.dwChannels;\r
--\r
-- // Get sample rate and format information.\r
-- std::vector<unsigned int> rates;\r
-- if ( inCaps.dwChannels >= 2 ) {\r
-- if ( inCaps.dwFormats & WAVE_FORMAT_1S16 ) info.nativeFormats |= RTAUDIO_SINT16;\r
-- if ( inCaps.dwFormats & WAVE_FORMAT_2S16 ) info.nativeFormats |= RTAUDIO_SINT16;\r
-- if ( inCaps.dwFormats & WAVE_FORMAT_4S16 ) info.nativeFormats |= RTAUDIO_SINT16;\r
-- if ( inCaps.dwFormats & WAVE_FORMAT_96S16 ) info.nativeFormats |= RTAUDIO_SINT16;\r
-- if ( inCaps.dwFormats & WAVE_FORMAT_1S08 ) info.nativeFormats |= RTAUDIO_SINT8;\r
-- if ( inCaps.dwFormats & WAVE_FORMAT_2S08 ) info.nativeFormats |= RTAUDIO_SINT8;\r
-- if ( inCaps.dwFormats & WAVE_FORMAT_4S08 ) info.nativeFormats |= RTAUDIO_SINT8;\r
-- if ( inCaps.dwFormats & WAVE_FORMAT_96S08 ) info.nativeFormats |= RTAUDIO_SINT8;\r
--\r
-- if ( info.nativeFormats & RTAUDIO_SINT16 ) {\r
-- if ( inCaps.dwFormats & WAVE_FORMAT_1S16 ) rates.push_back( 11025 );\r
-- if ( inCaps.dwFormats & WAVE_FORMAT_2S16 ) rates.push_back( 22050 );\r
-- if ( inCaps.dwFormats & WAVE_FORMAT_4S16 ) rates.push_back( 44100 );\r
-- if ( inCaps.dwFormats & WAVE_FORMAT_96S16 ) rates.push_back( 96000 );\r
-- }\r
-- else if ( info.nativeFormats & RTAUDIO_SINT8 ) {\r
-- if ( inCaps.dwFormats & WAVE_FORMAT_1S08 ) rates.push_back( 11025 );\r
-- if ( inCaps.dwFormats & WAVE_FORMAT_2S08 ) rates.push_back( 22050 );\r
-- if ( inCaps.dwFormats & WAVE_FORMAT_4S08 ) rates.push_back( 44100 );\r
-- if ( inCaps.dwFormats & WAVE_FORMAT_96S08 ) rates.push_back( 96000 );\r
-- }\r
-- }\r
-- else if ( inCaps.dwChannels == 1 ) {\r
-- if ( inCaps.dwFormats & WAVE_FORMAT_1M16 ) info.nativeFormats |= RTAUDIO_SINT16;\r
-- if ( inCaps.dwFormats & WAVE_FORMAT_2M16 ) info.nativeFormats |= RTAUDIO_SINT16;\r
-- if ( inCaps.dwFormats & WAVE_FORMAT_4M16 ) info.nativeFormats |= RTAUDIO_SINT16;\r
-- if ( inCaps.dwFormats & WAVE_FORMAT_96M16 ) info.nativeFormats |= RTAUDIO_SINT16;\r
-- if ( inCaps.dwFormats & WAVE_FORMAT_1M08 ) info.nativeFormats |= RTAUDIO_SINT8;\r
-- if ( inCaps.dwFormats & WAVE_FORMAT_2M08 ) info.nativeFormats |= RTAUDIO_SINT8;\r
-- if ( inCaps.dwFormats & WAVE_FORMAT_4M08 ) info.nativeFormats |= RTAUDIO_SINT8;\r
-- if ( inCaps.dwFormats & WAVE_FORMAT_96M08 ) info.nativeFormats |= RTAUDIO_SINT8;\r
--\r
-- if ( info.nativeFormats & RTAUDIO_SINT16 ) {\r
-- if ( inCaps.dwFormats & WAVE_FORMAT_1M16 ) rates.push_back( 11025 );\r
-- if ( inCaps.dwFormats & WAVE_FORMAT_2M16 ) rates.push_back( 22050 );\r
-- if ( inCaps.dwFormats & WAVE_FORMAT_4M16 ) rates.push_back( 44100 );\r
-- if ( inCaps.dwFormats & WAVE_FORMAT_96M16 ) rates.push_back( 96000 );\r
-- }\r
-- else if ( info.nativeFormats & RTAUDIO_SINT8 ) {\r
-- if ( inCaps.dwFormats & WAVE_FORMAT_1M08 ) rates.push_back( 11025 );\r
-- if ( inCaps.dwFormats & WAVE_FORMAT_2M08 ) rates.push_back( 22050 );\r
-- if ( inCaps.dwFormats & WAVE_FORMAT_4M08 ) rates.push_back( 44100 );\r
-- if ( inCaps.dwFormats & WAVE_FORMAT_96M08 ) rates.push_back( 96000 );\r
-- }\r
-- }\r
-- else info.inputChannels = 0; // technically, this would be an error\r
--\r
-- input->Release();\r
--\r
-- if ( info.inputChannels == 0 ) return info;\r
--\r
-- // Copy the supported rates to the info structure but avoid duplication.\r
-- bool found;\r
-- for ( unsigned int i=0; i<rates.size(); i++ ) {\r
-- found = false;\r
-- for ( unsigned int j=0; j<info.sampleRates.size(); j++ ) {\r
-- if ( rates[i] == info.sampleRates[j] ) {\r
-- found = true;\r
-- break;\r
-- }\r
-- }\r
-- if ( found == false ) info.sampleRates.push_back( rates[i] );\r
-- }\r
-- std::sort( info.sampleRates.begin(), info.sampleRates.end() );\r
--\r
-- // If device opens for both playback and capture, we determine the channels.\r
-- if ( info.outputChannels > 0 && info.inputChannels > 0 )\r
-- info.duplexChannels = (info.outputChannels > info.inputChannels) ? info.inputChannels : info.outputChannels;\r
--\r
-- if ( device == 0 ) info.isDefaultInput = true;\r
--\r
-- // Copy name and return.\r
-- info.name = dsDevices[ device ].name;\r
-- info.probed = true;\r
-- return info;\r
--}\r
--\r
--bool RtApiDs :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,\r
-- unsigned int firstChannel, unsigned int sampleRate,\r
-- RtAudioFormat format, unsigned int *bufferSize,\r
-- RtAudio::StreamOptions *options )\r
--{\r
-- if ( channels + firstChannel > 2 ) {\r
-- errorText_ = "RtApiDs::probeDeviceOpen: DirectSound does not support more than 2 channels per device.";\r
-- return FAILURE;\r
-- }\r
--\r
-- size_t nDevices = dsDevices.size();\r
-- if ( nDevices == 0 ) {\r
-- // This should not happen because a check is made before this function is called.\r
-- errorText_ = "RtApiDs::probeDeviceOpen: no devices found!";\r
-- return FAILURE;\r
-- }\r
--\r
-- if ( device >= nDevices ) {\r
-- // This should not happen because a check is made before this function is called.\r
-- errorText_ = "RtApiDs::probeDeviceOpen: device ID is invalid!";\r
-- return FAILURE;\r
-- }\r
--\r
-- if ( mode == OUTPUT ) {\r
-- if ( dsDevices[ device ].validId[0] == false ) {\r
-- errorStream_ << "RtApiDs::probeDeviceOpen: device (" << device << ") does not support output!";\r
-- errorText_ = errorStream_.str();\r
-- return FAILURE;\r
-- }\r
-- }\r
-- else { // mode == INPUT\r
-- if ( dsDevices[ device ].validId[1] == false ) {\r
-- errorStream_ << "RtApiDs::probeDeviceOpen: device (" << device << ") does not support input!";\r
-- errorText_ = errorStream_.str();\r
-- return FAILURE;\r
-- }\r
-- }\r
--\r
-- // According to a note in PortAudio, using GetDesktopWindow()\r
-- // instead of GetForegroundWindow() is supposed to avoid problems\r
-- // that occur when the application's window is not the foreground\r
-- // window. Also, if the application window closes before the\r
-- // DirectSound buffer, DirectSound can crash. In the past, I had\r
-- // problems when using GetDesktopWindow() but it seems fine now\r
-- // (January 2010). I'll leave it commented here.\r
-- // HWND hWnd = GetForegroundWindow();\r
-- HWND hWnd = GetDesktopWindow();\r
--\r
-- // Check the numberOfBuffers parameter and limit the lowest value to\r
-- // two. This is a judgement call and a value of two is probably too\r
-- // low for capture, but it should work for playback.\r
-- int nBuffers = 0;\r
-- if ( options ) nBuffers = options->numberOfBuffers;\r
-- if ( options && options->flags & RTAUDIO_MINIMIZE_LATENCY ) nBuffers = 2;\r
-- if ( nBuffers < 2 ) nBuffers = 3;\r
--\r
-- // Check the lower range of the user-specified buffer size and set\r
-- // (arbitrarily) to a lower bound of 32.\r
-- if ( *bufferSize < 32 ) *bufferSize = 32;\r
--\r
-- // Create the wave format structure. The data format setting will\r
-- // be determined later.\r
-- WAVEFORMATEX waveFormat;\r
-- ZeroMemory( &waveFormat, sizeof(WAVEFORMATEX) );\r
-- waveFormat.wFormatTag = WAVE_FORMAT_PCM;\r
-- waveFormat.nChannels = channels + firstChannel;\r
-- waveFormat.nSamplesPerSec = (unsigned long) sampleRate;\r
--\r
-- // Determine the device buffer size. By default, we'll use the value\r
-- // defined above (32K), but we will grow it to make allowances for\r
-- // very large software buffer sizes.\r
-- DWORD dsBufferSize = MINIMUM_DEVICE_BUFFER_SIZE;\r
-- DWORD dsPointerLeadTime = 0;\r
--\r
-- void *ohandle = 0, *bhandle = 0;\r
-- HRESULT result;\r
-- if ( mode == OUTPUT ) {\r
--\r
-- LPDIRECTSOUND output;\r
-- result = DirectSoundCreate( dsDevices[ device ].id[0], &output, NULL );\r
-- if ( FAILED( result ) ) {\r
-- errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") opening output device (" << dsDevices[ device ].name << ")!";\r
-- errorText_ = errorStream_.str();\r
-- return FAILURE;\r
-- }\r
--\r
-- DSCAPS outCaps;\r
-- outCaps.dwSize = sizeof( outCaps );\r
-- result = output->GetCaps( &outCaps );\r
-- if ( FAILED( result ) ) {\r
-- output->Release();\r
-- errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") getting capabilities (" << dsDevices[ device ].name << ")!";\r
-- errorText_ = errorStream_.str();\r
-- return FAILURE;\r
-- }\r
--\r
-- // Check channel information.\r
-- if ( channels + firstChannel == 2 && !( outCaps.dwFlags & DSCAPS_PRIMARYSTEREO ) ) {\r
-- errorStream_ << "RtApiDs::getDeviceInfo: the output device (" << dsDevices[ device ].name << ") does not support stereo playback.";\r
-- errorText_ = errorStream_.str();\r
-- return FAILURE;\r
-- }\r
--\r
-- // Check format information. Use 16-bit format unless not\r
-- // supported or user requests 8-bit.\r
-- if ( outCaps.dwFlags & DSCAPS_PRIMARY16BIT &&\r
-- !( format == RTAUDIO_SINT8 && outCaps.dwFlags & DSCAPS_PRIMARY8BIT ) ) {\r
-- waveFormat.wBitsPerSample = 16;\r
-- stream_.deviceFormat[mode] = RTAUDIO_SINT16;\r
-- }\r
-- else {\r
-- waveFormat.wBitsPerSample = 8;\r
-- stream_.deviceFormat[mode] = RTAUDIO_SINT8;\r
-- }\r
-- stream_.userFormat = format;\r
--\r
-- // Update wave format structure and buffer information.\r
-- waveFormat.nBlockAlign = waveFormat.nChannels * waveFormat.wBitsPerSample / 8;\r
-- waveFormat.nAvgBytesPerSec = waveFormat.nSamplesPerSec * waveFormat.nBlockAlign;\r
-- dsPointerLeadTime = nBuffers * (*bufferSize) * (waveFormat.wBitsPerSample / 8) * channels;\r
--\r
-- // If the user wants an even bigger buffer, increase the device buffer size accordingly.\r
-- while ( dsPointerLeadTime * 2U > dsBufferSize )\r
-- dsBufferSize *= 2;\r
--\r
-- // Set cooperative level to DSSCL_EXCLUSIVE ... sound stops when window focus changes.\r
-- // result = output->SetCooperativeLevel( hWnd, DSSCL_EXCLUSIVE );\r
-- // Set cooperative level to DSSCL_PRIORITY ... sound remains when window focus changes.\r
-- result = output->SetCooperativeLevel( hWnd, DSSCL_PRIORITY );\r
-- if ( FAILED( result ) ) {\r
-- output->Release();\r
-- errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") setting cooperative level (" << dsDevices[ device ].name << ")!";\r
-- errorText_ = errorStream_.str();\r
-- return FAILURE;\r
-- }\r
--\r
-- // Even though we will write to the secondary buffer, we need to\r
-- // access the primary buffer to set the correct output format\r
-- // (since the default is 8-bit, 22 kHz!). Setup the DS primary\r
-- // buffer description.\r
-- DSBUFFERDESC bufferDescription;\r
-- ZeroMemory( &bufferDescription, sizeof( DSBUFFERDESC ) );\r
-- bufferDescription.dwSize = sizeof( DSBUFFERDESC );\r
-- bufferDescription.dwFlags = DSBCAPS_PRIMARYBUFFER;\r
--\r
-- // Obtain the primary buffer\r
-- LPDIRECTSOUNDBUFFER buffer;\r
-- result = output->CreateSoundBuffer( &bufferDescription, &buffer, NULL );\r
-- if ( FAILED( result ) ) {\r
-- output->Release();\r
-- errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") accessing primary buffer (" << dsDevices[ device ].name << ")!";\r
-- errorText_ = errorStream_.str();\r
-- return FAILURE;\r
-- }\r
--\r
-- // Set the primary DS buffer sound format.\r
-- result = buffer->SetFormat( &waveFormat );\r
-- if ( FAILED( result ) ) {\r
-- output->Release();\r
-- errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") setting primary buffer format (" << dsDevices[ device ].name << ")!";\r
-- errorText_ = errorStream_.str();\r
-- return FAILURE;\r
-- }\r
--\r
-- // Setup the secondary DS buffer description.\r
-- ZeroMemory( &bufferDescription, sizeof( DSBUFFERDESC ) );\r
-- bufferDescription.dwSize = sizeof( DSBUFFERDESC );\r
-- bufferDescription.dwFlags = ( DSBCAPS_STICKYFOCUS |\r
-- DSBCAPS_GLOBALFOCUS |\r
-- DSBCAPS_GETCURRENTPOSITION2 |\r
-- DSBCAPS_LOCHARDWARE ); // Force hardware mixing\r
-- bufferDescription.dwBufferBytes = dsBufferSize;\r
-- bufferDescription.lpwfxFormat = &waveFormat;\r
--\r
-- // Try to create the secondary DS buffer. If that doesn't work,\r
-- // try to use software mixing. Otherwise, there's a problem.\r
-- result = output->CreateSoundBuffer( &bufferDescription, &buffer, NULL );\r
-- if ( FAILED( result ) ) {\r
-- bufferDescription.dwFlags = ( DSBCAPS_STICKYFOCUS |\r
-- DSBCAPS_GLOBALFOCUS |\r
-- DSBCAPS_GETCURRENTPOSITION2 |\r
-- DSBCAPS_LOCSOFTWARE ); // Force software mixing\r
-- result = output->CreateSoundBuffer( &bufferDescription, &buffer, NULL );\r
-- if ( FAILED( result ) ) {\r
-- output->Release();\r
-- errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") creating secondary buffer (" << dsDevices[ device ].name << ")!";\r
-- errorText_ = errorStream_.str();\r
-- return FAILURE;\r
-- }\r
-- }\r
--\r
-- // Get the buffer size ... might be different from what we specified.\r
-- DSBCAPS dsbcaps;\r
-- dsbcaps.dwSize = sizeof( DSBCAPS );\r
-- result = buffer->GetCaps( &dsbcaps );\r
-- if ( FAILED( result ) ) {\r
-- output->Release();\r
-- buffer->Release();\r
-- errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") getting buffer settings (" << dsDevices[ device ].name << ")!";\r
-- errorText_ = errorStream_.str();\r
-- return FAILURE;\r
-- }\r
--\r
-- dsBufferSize = dsbcaps.dwBufferBytes;\r
--\r
-- // Lock the DS buffer\r
-- LPVOID audioPtr;\r
-- DWORD dataLen;\r
-- result = buffer->Lock( 0, dsBufferSize, &audioPtr, &dataLen, NULL, NULL, 0 );\r
-- if ( FAILED( result ) ) {\r
-- output->Release();\r
-- buffer->Release();\r
-- errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") locking buffer (" << dsDevices[ device ].name << ")!";\r
-- errorText_ = errorStream_.str();\r
-- return FAILURE;\r
-- }\r
--\r
-- // Zero the DS buffer\r
-- ZeroMemory( audioPtr, dataLen );\r
--\r
-- // Unlock the DS buffer\r
-- result = buffer->Unlock( audioPtr, dataLen, NULL, 0 );\r
-- if ( FAILED( result ) ) {\r
-- output->Release();\r
-- buffer->Release();\r
-- errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") unlocking buffer (" << dsDevices[ device ].name << ")!";\r
-- errorText_ = errorStream_.str();\r
-- return FAILURE;\r
-- }\r
--\r
-- ohandle = (void *) output;\r
-- bhandle = (void *) buffer;\r
-- }\r
--\r
-- if ( mode == INPUT ) {\r
--\r
-- LPDIRECTSOUNDCAPTURE input;\r
-- result = DirectSoundCaptureCreate( dsDevices[ device ].id[1], &input, NULL );\r
-- if ( FAILED( result ) ) {\r
-- errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") opening input device (" << dsDevices[ device ].name << ")!";\r
-- errorText_ = errorStream_.str();\r
-- return FAILURE;\r
-- }\r
--\r
-- DSCCAPS inCaps;\r
-- inCaps.dwSize = sizeof( inCaps );\r
-- result = input->GetCaps( &inCaps );\r
-- if ( FAILED( result ) ) {\r
-- input->Release();\r
-- errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") getting input capabilities (" << dsDevices[ device ].name << ")!";\r
-- errorText_ = errorStream_.str();\r
-- return FAILURE;\r
-- }\r
--\r
-- // Check channel information.\r
-- if ( inCaps.dwChannels < channels + firstChannel ) {\r
-- errorText_ = "RtApiDs::getDeviceInfo: the input device does not support requested input channels.";\r
-- return FAILURE;\r
-- }\r
--\r
-- // Check format information. Use 16-bit format unless user\r
-- // requests 8-bit.\r
-- DWORD deviceFormats;\r
-- if ( channels + firstChannel == 2 ) {\r
-- deviceFormats = WAVE_FORMAT_1S08 | WAVE_FORMAT_2S08 | WAVE_FORMAT_4S08 | WAVE_FORMAT_96S08;\r
-- if ( format == RTAUDIO_SINT8 && inCaps.dwFormats & deviceFormats ) {\r
-- waveFormat.wBitsPerSample = 8;\r
-- stream_.deviceFormat[mode] = RTAUDIO_SINT8;\r
-- }\r
-- else { // assume 16-bit is supported\r
-- waveFormat.wBitsPerSample = 16;\r
-- stream_.deviceFormat[mode] = RTAUDIO_SINT16;\r
-- }\r
-- }\r
-- else { // channel == 1\r
-- deviceFormats = WAVE_FORMAT_1M08 | WAVE_FORMAT_2M08 | WAVE_FORMAT_4M08 | WAVE_FORMAT_96M08;\r
-- if ( format == RTAUDIO_SINT8 && inCaps.dwFormats & deviceFormats ) {\r
-- waveFormat.wBitsPerSample = 8;\r
-- stream_.deviceFormat[mode] = RTAUDIO_SINT8;\r
-- }\r
-- else { // assume 16-bit is supported\r
-- waveFormat.wBitsPerSample = 16;\r
-- stream_.deviceFormat[mode] = RTAUDIO_SINT16;\r
-- }\r
-- }\r
-- stream_.userFormat = format;\r
--\r
-- // Update wave format structure and buffer information.\r
-- waveFormat.nBlockAlign = waveFormat.nChannels * waveFormat.wBitsPerSample / 8;\r
-- waveFormat.nAvgBytesPerSec = waveFormat.nSamplesPerSec * waveFormat.nBlockAlign;\r
-- dsPointerLeadTime = nBuffers * (*bufferSize) * (waveFormat.wBitsPerSample / 8) * channels;\r
--\r
-- // If the user wants an even bigger buffer, increase the device buffer size accordingly.\r
-- while ( dsPointerLeadTime * 2U > dsBufferSize )\r
-- dsBufferSize *= 2;\r
--\r
-- // Setup the secondary DS buffer description.\r
-- DSCBUFFERDESC bufferDescription;\r
-- ZeroMemory( &bufferDescription, sizeof( DSCBUFFERDESC ) );\r
-- bufferDescription.dwSize = sizeof( DSCBUFFERDESC );\r
-- bufferDescription.dwFlags = 0;\r
-- bufferDescription.dwReserved = 0;\r
-- bufferDescription.dwBufferBytes = dsBufferSize;\r
-- bufferDescription.lpwfxFormat = &waveFormat;\r
--\r
-- // Create the capture buffer.\r
-- LPDIRECTSOUNDCAPTUREBUFFER buffer;\r
-- result = input->CreateCaptureBuffer( &bufferDescription, &buffer, NULL );\r
-- if ( FAILED( result ) ) {\r
-- input->Release();\r
-- errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") creating input buffer (" << dsDevices[ device ].name << ")!";\r
-- errorText_ = errorStream_.str();\r
-- return FAILURE;\r
-- }\r
--\r
-- // Get the buffer size ... might be different from what we specified.\r
-- DSCBCAPS dscbcaps;\r
-- dscbcaps.dwSize = sizeof( DSCBCAPS );\r
-- result = buffer->GetCaps( &dscbcaps );\r
-- if ( FAILED( result ) ) {\r
-- input->Release();\r
-- buffer->Release();\r
-- errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") getting buffer settings (" << dsDevices[ device ].name << ")!";\r
-- errorText_ = errorStream_.str();\r
-- return FAILURE;\r
-- }\r
--\r
-- dsBufferSize = dscbcaps.dwBufferBytes;\r
--\r
-- // NOTE: We could have a problem here if this is a duplex stream\r
-- // and the play and capture hardware buffer sizes are different\r
-- // (I'm actually not sure if that is a problem or not).\r
-- // Currently, we are not verifying that.\r
--\r
-- // Lock the capture buffer\r
-- LPVOID audioPtr;\r
-- DWORD dataLen;\r
-- result = buffer->Lock( 0, dsBufferSize, &audioPtr, &dataLen, NULL, NULL, 0 );\r
-- if ( FAILED( result ) ) {\r
-- input->Release();\r
-- buffer->Release();\r
-- errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") locking input buffer (" << dsDevices[ device ].name << ")!";\r
-- errorText_ = errorStream_.str();\r
-- return FAILURE;\r
-- }\r
--\r
-- // Zero the buffer\r
-- ZeroMemory( audioPtr, dataLen );\r
--\r
-- // Unlock the buffer\r
-- result = buffer->Unlock( audioPtr, dataLen, NULL, 0 );\r
-- if ( FAILED( result ) ) {\r
-- input->Release();\r
-- buffer->Release();\r
-- errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") unlocking input buffer (" << dsDevices[ device ].name << ")!";\r
-- errorText_ = errorStream_.str();\r
-- return FAILURE;\r
-- }\r
--\r
-- ohandle = (void *) input;\r
-- bhandle = (void *) buffer;\r
-- }\r
--\r
-- // Set various stream parameters\r
-- DsHandle *handle = 0;\r
-- stream_.nDeviceChannels[mode] = channels + firstChannel;\r
-- stream_.nUserChannels[mode] = channels;\r
-- stream_.bufferSize = *bufferSize;\r
-- stream_.channelOffset[mode] = firstChannel;\r
-- stream_.deviceInterleaved[mode] = true;\r
-- if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) stream_.userInterleaved = false;\r
-- else stream_.userInterleaved = true;\r
--\r
-- // Set flag for buffer conversion\r
-- stream_.doConvertBuffer[mode] = false;\r
-- if (stream_.nUserChannels[mode] != stream_.nDeviceChannels[mode])\r
-- stream_.doConvertBuffer[mode] = true;\r
-- if (stream_.userFormat != stream_.deviceFormat[mode])\r
-- stream_.doConvertBuffer[mode] = true;\r
-- if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] &&\r
-- stream_.nUserChannels[mode] > 1 )\r
-- stream_.doConvertBuffer[mode] = true;\r
--\r
-- // Allocate necessary internal buffers\r
-- long bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat );\r
-- stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 );\r
-- if ( stream_.userBuffer[mode] == NULL ) {\r
-- errorText_ = "RtApiDs::probeDeviceOpen: error allocating user buffer memory.";\r
-- goto error;\r
-- }\r
--\r
-- if ( stream_.doConvertBuffer[mode] ) {\r
--\r
-- bool makeBuffer = true;\r
-- bufferBytes = stream_.nDeviceChannels[mode] * formatBytes( stream_.deviceFormat[mode] );\r
-- if ( mode == INPUT ) {\r
-- if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) {\r
-- unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] );\r
-- if ( bufferBytes <= (long) bytesOut ) makeBuffer = false;\r
-- }\r
-- }\r
--\r
-- if ( makeBuffer ) {\r
-- bufferBytes *= *bufferSize;\r
-- if ( stream_.deviceBuffer ) free( stream_.deviceBuffer );\r
-- stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 );\r
-- if ( stream_.deviceBuffer == NULL ) {\r
-- errorText_ = "RtApiDs::probeDeviceOpen: error allocating device buffer memory.";\r
-- goto error;\r
-- }\r
-- }\r
-- }\r
--\r
-- // Allocate our DsHandle structures for the stream.\r
-- if ( stream_.apiHandle == 0 ) {\r
-- try {\r
-- handle = new DsHandle;\r
-- }\r
-- catch ( std::bad_alloc& ) {\r
-- errorText_ = "RtApiDs::probeDeviceOpen: error allocating AsioHandle memory.";\r
-- goto error;\r
-- }\r
--\r
-- // Create a manual-reset event.\r
-- handle->condition = CreateEvent( NULL, // no security\r
-- TRUE, // manual-reset\r
-- FALSE, // non-signaled initially\r
-- NULL ); // unnamed\r
-- stream_.apiHandle = (void *) handle;\r
-- }\r
-- else\r
-- handle = (DsHandle *) stream_.apiHandle;\r
-- handle->id[mode] = ohandle;\r
-- handle->buffer[mode] = bhandle;\r
-- handle->dsBufferSize[mode] = dsBufferSize;\r
-- handle->dsPointerLeadTime[mode] = dsPointerLeadTime;\r
--\r
-- stream_.device[mode] = device;\r
-- stream_.state = STREAM_STOPPED;\r
-- if ( stream_.mode == OUTPUT && mode == INPUT )\r
-- // We had already set up an output stream.\r
-- stream_.mode = DUPLEX;\r
-- else\r
-- stream_.mode = mode;\r
-- stream_.nBuffers = nBuffers;\r
-- stream_.sampleRate = sampleRate;\r
--\r
-- // Setup the buffer conversion information structure.\r
-- if ( stream_.doConvertBuffer[mode] ) setConvertInfo( mode, firstChannel );\r
--\r
-- // Setup the callback thread.\r
-- if ( stream_.callbackInfo.isRunning == false ) {\r
-- unsigned threadId;\r
-- stream_.callbackInfo.isRunning = true;\r
-- stream_.callbackInfo.object = (void *) this;\r
-- stream_.callbackInfo.thread = _beginthreadex( NULL, 0, &callbackHandler,\r
-- &stream_.callbackInfo, 0, &threadId );\r
-- if ( stream_.callbackInfo.thread == 0 ) {\r
-- errorText_ = "RtApiDs::probeDeviceOpen: error creating callback thread!";\r
-- goto error;\r
-- }\r
--\r
-- // Boost DS thread priority\r
-- SetThreadPriority( (HANDLE) stream_.callbackInfo.thread, THREAD_PRIORITY_HIGHEST );\r
-- }\r
-- return SUCCESS;\r
--\r
-- error:\r
-- if ( handle ) {\r
-- if ( handle->buffer[0] ) { // the object pointer can be NULL and valid\r
-- LPDIRECTSOUND object = (LPDIRECTSOUND) handle->id[0];\r
-- LPDIRECTSOUNDBUFFER buffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0];\r
-- if ( buffer ) buffer->Release();\r
-- object->Release();\r
-- }\r
-- if ( handle->buffer[1] ) {\r
-- LPDIRECTSOUNDCAPTURE object = (LPDIRECTSOUNDCAPTURE) handle->id[1];\r
-- LPDIRECTSOUNDCAPTUREBUFFER buffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1];\r
-- if ( buffer ) buffer->Release();\r
-- object->Release();\r
-- }\r
-- CloseHandle( handle->condition );\r
-- delete handle;\r
-- stream_.apiHandle = 0;\r
-- }\r
--\r
-- for ( int i=0; i<2; i++ ) {\r
-- if ( stream_.userBuffer[i] ) {\r
-- free( stream_.userBuffer[i] );\r
-- stream_.userBuffer[i] = 0;\r
-- }\r
-- }\r
--\r
-- if ( stream_.deviceBuffer ) {\r
-- free( stream_.deviceBuffer );\r
-- stream_.deviceBuffer = 0;\r
-- }\r
--\r
-- stream_.state = STREAM_CLOSED;\r
-- return FAILURE;\r
--}\r
--\r
--void RtApiDs :: closeStream()\r
--{\r
-- if ( stream_.state == STREAM_CLOSED ) {\r
-- errorText_ = "RtApiDs::closeStream(): no open stream to close!";\r
-- error( RtAudioError::WARNING );\r
-- return;\r
-- }\r
--\r
-- // Stop the callback thread.\r
-- stream_.callbackInfo.isRunning = false;\r
-- WaitForSingleObject( (HANDLE) stream_.callbackInfo.thread, INFINITE );\r
-- CloseHandle( (HANDLE) stream_.callbackInfo.thread );\r
--\r
-- DsHandle *handle = (DsHandle *) stream_.apiHandle;\r
-- if ( handle ) {\r
-- if ( handle->buffer[0] ) { // the object pointer can be NULL and valid\r
-- LPDIRECTSOUND object = (LPDIRECTSOUND) handle->id[0];\r
-- LPDIRECTSOUNDBUFFER buffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0];\r
-- if ( buffer ) {\r
-- buffer->Stop();\r
-- buffer->Release();\r
-- }\r
-- object->Release();\r
-- }\r
-- if ( handle->buffer[1] ) {\r
-- LPDIRECTSOUNDCAPTURE object = (LPDIRECTSOUNDCAPTURE) handle->id[1];\r
-- LPDIRECTSOUNDCAPTUREBUFFER buffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1];\r
-- if ( buffer ) {\r
-- buffer->Stop();\r
-- buffer->Release();\r
-- }\r
-- object->Release();\r
-- }\r
-- CloseHandle( handle->condition );\r
-- delete handle;\r
-- stream_.apiHandle = 0;\r
-- }\r
--\r
-- for ( int i=0; i<2; i++ ) {\r
-- if ( stream_.userBuffer[i] ) {\r
-- free( stream_.userBuffer[i] );\r
-- stream_.userBuffer[i] = 0;\r
-- }\r
-- }\r
--\r
-- if ( stream_.deviceBuffer ) {\r
-- free( stream_.deviceBuffer );\r
-- stream_.deviceBuffer = 0;\r
-- }\r
--\r
-- stream_.mode = UNINITIALIZED;\r
-- stream_.state = STREAM_CLOSED;\r
--}\r
--\r
--void RtApiDs :: startStream()\r
--{\r
-- verifyStream();\r
-- if ( stream_.state == STREAM_RUNNING ) {\r
-- errorText_ = "RtApiDs::startStream(): the stream is already running!";\r
-- error( RtAudioError::WARNING );\r
-- return;\r
-- }\r
--\r
-- DsHandle *handle = (DsHandle *) stream_.apiHandle;\r
--\r
-- // Increase scheduler frequency on lesser windows (a side-effect of\r
-- // increasing timer accuracy). On greater windows (Win2K or later),\r
-- // this is already in effect.\r
-- timeBeginPeriod( 1 );\r
--\r
-- buffersRolling = false;\r
-- duplexPrerollBytes = 0;\r
--\r
-- if ( stream_.mode == DUPLEX ) {\r
-- // 0.5 seconds of silence in DUPLEX mode while the devices spin up and synchronize.\r
-- duplexPrerollBytes = (int) ( 0.5 * stream_.sampleRate * formatBytes( stream_.deviceFormat[1] ) * stream_.nDeviceChannels[1] );\r
-- }\r
--\r
-- HRESULT result = 0;\r
-- if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {\r
--\r
-- LPDIRECTSOUNDBUFFER buffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0];\r
-- result = buffer->Play( 0, 0, DSBPLAY_LOOPING );\r
-- if ( FAILED( result ) ) {\r
-- errorStream_ << "RtApiDs::startStream: error (" << getErrorString( result ) << ") starting output buffer!";\r
-- errorText_ = errorStream_.str();\r
-- goto unlock;\r
-- }\r
-- }\r
--\r
-- if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {\r
--\r
-- LPDIRECTSOUNDCAPTUREBUFFER buffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1];\r
-- result = buffer->Start( DSCBSTART_LOOPING );\r
-- if ( FAILED( result ) ) {\r
-- errorStream_ << "RtApiDs::startStream: error (" << getErrorString( result ) << ") starting input buffer!";\r
-- errorText_ = errorStream_.str();\r
-- goto unlock;\r
-- }\r
-- }\r
--\r
-- handle->drainCounter = 0;\r
-- handle->internalDrain = false;\r
-- ResetEvent( handle->condition );\r
-- stream_.state = STREAM_RUNNING;\r
--\r
-- unlock:\r
-- if ( FAILED( result ) ) error( RtAudioError::SYSTEM_ERROR );\r
--}\r
--\r
--void RtApiDs :: stopStream()\r
--{\r
-- verifyStream();\r
-- if ( stream_.state == STREAM_STOPPED ) {\r
-- errorText_ = "RtApiDs::stopStream(): the stream is already stopped!";\r
-- error( RtAudioError::WARNING );\r
-- return;\r
-- }\r
--\r
-- HRESULT result = 0;\r
-- LPVOID audioPtr;\r
-- DWORD dataLen;\r
-- DsHandle *handle = (DsHandle *) stream_.apiHandle;\r
-- if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {\r
-- if ( handle->drainCounter == 0 ) {\r
-- handle->drainCounter = 2;\r
-- WaitForSingleObject( handle->condition, INFINITE ); // block until signaled\r
-- }\r
--\r
-- stream_.state = STREAM_STOPPED;\r
--\r
-- MUTEX_LOCK( &stream_.mutex );\r
--\r
-- // Stop the buffer and clear memory\r
-- LPDIRECTSOUNDBUFFER buffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0];\r
-- result = buffer->Stop();\r
-- if ( FAILED( result ) ) {\r
-- errorStream_ << "RtApiDs::stopStream: error (" << getErrorString( result ) << ") stopping output buffer!";\r
-- errorText_ = errorStream_.str();\r
-- goto unlock;\r
-- }\r
--\r
-- // Lock the buffer and clear it so that if we start to play again,\r
-- // we won't have old data playing.\r
-- result = buffer->Lock( 0, handle->dsBufferSize[0], &audioPtr, &dataLen, NULL, NULL, 0 );\r
-- if ( FAILED( result ) ) {\r
-- errorStream_ << "RtApiDs::stopStream: error (" << getErrorString( result ) << ") locking output buffer!";\r
-- errorText_ = errorStream_.str();\r
-- goto unlock;\r
-- }\r
--\r
-- // Zero the DS buffer\r
-- ZeroMemory( audioPtr, dataLen );\r
--\r
-- // Unlock the DS buffer\r
-- result = buffer->Unlock( audioPtr, dataLen, NULL, 0 );\r
-- if ( FAILED( result ) ) {\r
-- errorStream_ << "RtApiDs::stopStream: error (" << getErrorString( result ) << ") unlocking output buffer!";\r
-- errorText_ = errorStream_.str();\r
-- goto unlock;\r
-- }\r
--\r
-- // If we start playing again, we must begin at beginning of buffer.\r
-- handle->bufferPointer[0] = 0;\r
-- }\r
--\r
-- if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {\r
-- LPDIRECTSOUNDCAPTUREBUFFER buffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1];\r
-- audioPtr = NULL;\r
-- dataLen = 0;\r
--\r
-- stream_.state = STREAM_STOPPED;\r
--\r
-- if ( stream_.mode != DUPLEX )\r
-- MUTEX_LOCK( &stream_.mutex );\r
--\r
-- result = buffer->Stop();\r
-- if ( FAILED( result ) ) {\r
-- errorStream_ << "RtApiDs::stopStream: error (" << getErrorString( result ) << ") stopping input buffer!";\r
-- errorText_ = errorStream_.str();\r
-- goto unlock;\r
-- }\r
--\r
-- // Lock the buffer and clear it so that if we start to play again,\r
-- // we won't have old data playing.\r
-- result = buffer->Lock( 0, handle->dsBufferSize[1], &audioPtr, &dataLen, NULL, NULL, 0 );\r
-- if ( FAILED( result ) ) {\r
-- errorStream_ << "RtApiDs::stopStream: error (" << getErrorString( result ) << ") locking input buffer!";\r
-- errorText_ = errorStream_.str();\r
-- goto unlock;\r
-- }\r
--\r
-- // Zero the DS buffer\r
-- ZeroMemory( audioPtr, dataLen );\r
--\r
-- // Unlock the DS buffer\r
-- result = buffer->Unlock( audioPtr, dataLen, NULL, 0 );\r
-- if ( FAILED( result ) ) {\r
-- errorStream_ << "RtApiDs::stopStream: error (" << getErrorString( result ) << ") unlocking input buffer!";\r
-- errorText_ = errorStream_.str();\r
-- goto unlock;\r
-- }\r
--\r
-- // If we start recording again, we must begin at beginning of buffer.\r
-- handle->bufferPointer[1] = 0;\r
-- }\r
--\r
-- unlock:\r
-- timeEndPeriod( 1 ); // revert to normal scheduler frequency on lesser windows.\r
-- MUTEX_UNLOCK( &stream_.mutex );\r
--\r
-- if ( FAILED( result ) ) error( RtAudioError::SYSTEM_ERROR );\r
--}\r
--\r
--void RtApiDs :: abortStream()\r
--{\r
-- verifyStream();\r
-- if ( stream_.state == STREAM_STOPPED ) {\r
-- errorText_ = "RtApiDs::abortStream(): the stream is already stopped!";\r
-- error( RtAudioError::WARNING );\r
-- return;\r
-- }\r
--\r
-- DsHandle *handle = (DsHandle *) stream_.apiHandle;\r
-- handle->drainCounter = 2;\r
--\r
-- stopStream();\r
--}\r
--\r
--void RtApiDs :: callbackEvent()\r
--{\r
-- if ( stream_.state == STREAM_STOPPED || stream_.state == STREAM_STOPPING ) {\r
-- Sleep( 50 ); // sleep 50 milliseconds\r
-- return;\r
-- }\r
--\r
-- if ( stream_.state == STREAM_CLOSED ) {\r
-- errorText_ = "RtApiDs::callbackEvent(): the stream is closed ... this shouldn't happen!";\r
-- error( RtAudioError::WARNING );\r
-- return;\r
-- }\r
--\r
-- CallbackInfo *info = (CallbackInfo *) &stream_.callbackInfo;\r
-- DsHandle *handle = (DsHandle *) stream_.apiHandle;\r
--\r
-- // Check if we were draining the stream and signal is finished.\r
-- if ( handle->drainCounter > stream_.nBuffers + 2 ) {\r
--\r
-- stream_.state = STREAM_STOPPING;\r
-- if ( handle->internalDrain == false )\r
-- SetEvent( handle->condition );\r
-- else\r
-- stopStream();\r
-- return;\r
-- }\r
--\r
-- // Invoke user callback to get fresh output data UNLESS we are\r
-- // draining stream.\r
-- if ( handle->drainCounter == 0 ) {\r
-- RtAudioCallback callback = (RtAudioCallback) info->callback;\r
-- double streamTime = getStreamTime();\r
-- RtAudioStreamStatus status = 0;\r
-- if ( stream_.mode != INPUT && handle->xrun[0] == true ) {\r
-- status |= RTAUDIO_OUTPUT_UNDERFLOW;\r
-- handle->xrun[0] = false;\r
-- }\r
-- if ( stream_.mode != OUTPUT && handle->xrun[1] == true ) {\r
-- status |= RTAUDIO_INPUT_OVERFLOW;\r
-- handle->xrun[1] = false;\r
-- }\r
-- int cbReturnValue = callback( stream_.userBuffer[0], stream_.userBuffer[1],\r
-- stream_.bufferSize, streamTime, status, info->userData );\r
-- if ( cbReturnValue == 2 ) {\r
-- stream_.state = STREAM_STOPPING;\r
-- handle->drainCounter = 2;\r
-- abortStream();\r
-- return;\r
-- }\r
-- else if ( cbReturnValue == 1 ) {\r
-- handle->drainCounter = 1;\r
-- handle->internalDrain = true;\r
-- }\r
-- }\r
--\r
-- HRESULT result;\r
-- DWORD currentWritePointer, safeWritePointer;\r
-- DWORD currentReadPointer, safeReadPointer;\r
-- UINT nextWritePointer;\r
--\r
-- LPVOID buffer1 = NULL;\r
-- LPVOID buffer2 = NULL;\r
-- DWORD bufferSize1 = 0;\r
-- DWORD bufferSize2 = 0;\r
--\r
-- char *buffer;\r
-- long bufferBytes;\r
--\r
-- MUTEX_LOCK( &stream_.mutex );\r
-- if ( stream_.state == STREAM_STOPPED ) {\r
-- MUTEX_UNLOCK( &stream_.mutex );\r
-- return;\r
-- }\r
--\r
-- if ( buffersRolling == false ) {\r
-- if ( stream_.mode == DUPLEX ) {\r
-- //assert( handle->dsBufferSize[0] == handle->dsBufferSize[1] );\r
--\r
-- // It takes a while for the devices to get rolling. As a result,\r
-- // there's no guarantee that the capture and write device pointers\r
-- // will move in lockstep. Wait here for both devices to start\r
-- // rolling, and then set our buffer pointers accordingly.\r
-- // e.g. Crystal Drivers: the capture buffer starts up 5700 to 9600\r
-- // bytes later than the write buffer.\r
--\r
-- // Stub: a serious risk of having a pre-emptive scheduling round\r
-- // take place between the two GetCurrentPosition calls... but I'm\r
-- // really not sure how to solve the problem. Temporarily boost to\r
-- // Realtime priority, maybe; but I'm not sure what priority the\r
-- // DirectSound service threads run at. We *should* be roughly\r
-- // within a ms or so of correct.\r
--\r
-- LPDIRECTSOUNDBUFFER dsWriteBuffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0];\r
-- LPDIRECTSOUNDCAPTUREBUFFER dsCaptureBuffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1];\r
--\r
-- DWORD startSafeWritePointer, startSafeReadPointer;\r
--\r
-- result = dsWriteBuffer->GetCurrentPosition( NULL, &startSafeWritePointer );\r
-- if ( FAILED( result ) ) {\r
-- errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current write position!";\r
-- errorText_ = errorStream_.str();\r
-- MUTEX_UNLOCK( &stream_.mutex );\r
-- error( RtAudioError::SYSTEM_ERROR );\r
-- return;\r
-- }\r
-- result = dsCaptureBuffer->GetCurrentPosition( NULL, &startSafeReadPointer );\r
-- if ( FAILED( result ) ) {\r
-- errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current read position!";\r
-- errorText_ = errorStream_.str();\r
-- MUTEX_UNLOCK( &stream_.mutex );\r
-- error( RtAudioError::SYSTEM_ERROR );\r
-- return;\r
-- }\r
-- while ( true ) {\r
-- result = dsWriteBuffer->GetCurrentPosition( NULL, &safeWritePointer );\r
-- if ( FAILED( result ) ) {\r
-- errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current write position!";\r
-- errorText_ = errorStream_.str();\r
-- MUTEX_UNLOCK( &stream_.mutex );\r
-- error( RtAudioError::SYSTEM_ERROR );\r
-- return;\r
-- }\r
-- result = dsCaptureBuffer->GetCurrentPosition( NULL, &safeReadPointer );\r
-- if ( FAILED( result ) ) {\r
-- errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current read position!";\r
-- errorText_ = errorStream_.str();\r
-- MUTEX_UNLOCK( &stream_.mutex );\r
-- error( RtAudioError::SYSTEM_ERROR );\r
-- return;\r
-- }\r
-- if ( safeWritePointer != startSafeWritePointer && safeReadPointer != startSafeReadPointer ) break;\r
-- Sleep( 1 );\r
-- }\r
--\r
-- //assert( handle->dsBufferSize[0] == handle->dsBufferSize[1] );\r
--\r
-- handle->bufferPointer[0] = safeWritePointer + handle->dsPointerLeadTime[0];\r
-- if ( handle->bufferPointer[0] >= handle->dsBufferSize[0] ) handle->bufferPointer[0] -= handle->dsBufferSize[0];\r
-- handle->bufferPointer[1] = safeReadPointer;\r
-- }\r
-- else if ( stream_.mode == OUTPUT ) {\r
--\r
-- // Set the proper nextWritePosition after initial startup.\r
-- LPDIRECTSOUNDBUFFER dsWriteBuffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0];\r
-- result = dsWriteBuffer->GetCurrentPosition( ¤tWritePointer, &safeWritePointer );\r
-- if ( FAILED( result ) ) {\r
-- errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current write position!";\r
-- errorText_ = errorStream_.str();\r
-- MUTEX_UNLOCK( &stream_.mutex );\r
-- error( RtAudioError::SYSTEM_ERROR );\r
-- return;\r
-- }\r
-- handle->bufferPointer[0] = safeWritePointer + handle->dsPointerLeadTime[0];\r
-- if ( handle->bufferPointer[0] >= handle->dsBufferSize[0] ) handle->bufferPointer[0] -= handle->dsBufferSize[0];\r
-- }\r
--\r
-- buffersRolling = true;\r
-- }\r
--\r
-- if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {\r
--\r
-- LPDIRECTSOUNDBUFFER dsBuffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0];\r
--\r
-- if ( handle->drainCounter > 1 ) { // write zeros to the output stream\r
-- bufferBytes = stream_.bufferSize * stream_.nUserChannels[0];\r
-- bufferBytes *= formatBytes( stream_.userFormat );\r
-- memset( stream_.userBuffer[0], 0, bufferBytes );\r
-- }\r
--\r
-- // Setup parameters and do buffer conversion if necessary.\r
-- if ( stream_.doConvertBuffer[0] ) {\r
-- buffer = stream_.deviceBuffer;\r
-- convertBuffer( buffer, stream_.userBuffer[0], stream_.convertInfo[0] );\r
-- bufferBytes = stream_.bufferSize * stream_.nDeviceChannels[0];\r
-- bufferBytes *= formatBytes( stream_.deviceFormat[0] );\r
-- }\r
-- else {\r
-- buffer = stream_.userBuffer[0];\r
-- bufferBytes = stream_.bufferSize * stream_.nUserChannels[0];\r
-- bufferBytes *= formatBytes( stream_.userFormat );\r
-- }\r
--\r
-- // No byte swapping necessary in DirectSound implementation.\r
--\r
-- // Ahhh ... windoze. 16-bit data is signed but 8-bit data is\r
-- // unsigned. So, we need to convert our signed 8-bit data here to\r
-- // unsigned.\r
-- if ( stream_.deviceFormat[0] == RTAUDIO_SINT8 )\r
-- for ( int i=0; i<bufferBytes; i++ ) buffer[i] = (unsigned char) ( buffer[i] + 128 );\r
--\r
-- DWORD dsBufferSize = handle->dsBufferSize[0];\r
-- nextWritePointer = handle->bufferPointer[0];\r
--\r
-- DWORD endWrite, leadPointer;\r
-- while ( true ) {\r
-- // Find out where the read and "safe write" pointers are.\r
-- result = dsBuffer->GetCurrentPosition( ¤tWritePointer, &safeWritePointer );\r
-- if ( FAILED( result ) ) {\r
-- errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current write position!";\r
-- errorText_ = errorStream_.str();\r
-- MUTEX_UNLOCK( &stream_.mutex );\r
-- error( RtAudioError::SYSTEM_ERROR );\r
-- return;\r
-- }\r
--\r
-- // We will copy our output buffer into the region between\r
-- // safeWritePointer and leadPointer. If leadPointer is not\r
-- // beyond the next endWrite position, wait until it is.\r
-- leadPointer = safeWritePointer + handle->dsPointerLeadTime[0];\r
-- //std::cout << "safeWritePointer = " << safeWritePointer << ", leadPointer = " << leadPointer << ", nextWritePointer = " << nextWritePointer << std::endl;\r
-- if ( leadPointer > dsBufferSize ) leadPointer -= dsBufferSize;\r
-- if ( leadPointer < nextWritePointer ) leadPointer += dsBufferSize; // unwrap offset\r
-- endWrite = nextWritePointer + bufferBytes;\r
--\r
-- // Check whether the entire write region is behind the play pointer.\r
-- if ( leadPointer >= endWrite ) break;\r
--\r
-- // If we are here, then we must wait until the leadPointer advances\r
-- // beyond the end of our next write region. We use the\r
-- // Sleep() function to suspend operation until that happens.\r
-- double millis = ( endWrite - leadPointer ) * 1000.0;\r
-- millis /= ( formatBytes( stream_.deviceFormat[0]) * stream_.nDeviceChannels[0] * stream_.sampleRate);\r
-- if ( millis < 1.0 ) millis = 1.0;\r
-- Sleep( (DWORD) millis );\r
-- }\r
--\r
-- if ( dsPointerBetween( nextWritePointer, safeWritePointer, currentWritePointer, dsBufferSize )\r
-- || dsPointerBetween( endWrite, safeWritePointer, currentWritePointer, dsBufferSize ) ) {\r
-- // We've strayed into the forbidden zone ... resync the read pointer.\r
-- handle->xrun[0] = true;\r
-- nextWritePointer = safeWritePointer + handle->dsPointerLeadTime[0] - bufferBytes;\r
-- if ( nextWritePointer >= dsBufferSize ) nextWritePointer -= dsBufferSize;\r
-- handle->bufferPointer[0] = nextWritePointer;\r
-- endWrite = nextWritePointer + bufferBytes;\r
-- }\r
--\r
-- // Lock free space in the buffer\r
-- result = dsBuffer->Lock( nextWritePointer, bufferBytes, &buffer1,\r
-- &bufferSize1, &buffer2, &bufferSize2, 0 );\r
-- if ( FAILED( result ) ) {\r
-- errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") locking buffer during playback!";\r
-- errorText_ = errorStream_.str();\r
-- MUTEX_UNLOCK( &stream_.mutex );\r
-- error( RtAudioError::SYSTEM_ERROR );\r
-- return;\r
-- }\r
--\r
-- // Copy our buffer into the DS buffer\r
-- CopyMemory( buffer1, buffer, bufferSize1 );\r
-- if ( buffer2 != NULL ) CopyMemory( buffer2, buffer+bufferSize1, bufferSize2 );\r
--\r
-- // Update our buffer offset and unlock sound buffer\r
-- dsBuffer->Unlock( buffer1, bufferSize1, buffer2, bufferSize2 );\r
-- if ( FAILED( result ) ) {\r
-- errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") unlocking buffer during playback!";\r
-- errorText_ = errorStream_.str();\r
-- MUTEX_UNLOCK( &stream_.mutex );\r
-- error( RtAudioError::SYSTEM_ERROR );\r
-- return;\r
-- }\r
-- nextWritePointer = ( nextWritePointer + bufferSize1 + bufferSize2 ) % dsBufferSize;\r
-- handle->bufferPointer[0] = nextWritePointer;\r
-- }\r
--\r
-- // Don't bother draining input\r
-- if ( handle->drainCounter ) {\r
-- handle->drainCounter++;\r
-- goto unlock;\r
-- }\r
--\r
-- if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {\r
--\r
-- // Setup parameters.\r
-- if ( stream_.doConvertBuffer[1] ) {\r
-- buffer = stream_.deviceBuffer;\r
-- bufferBytes = stream_.bufferSize * stream_.nDeviceChannels[1];\r
-- bufferBytes *= formatBytes( stream_.deviceFormat[1] );\r
-- }\r
-- else {\r
-- buffer = stream_.userBuffer[1];\r
-- bufferBytes = stream_.bufferSize * stream_.nUserChannels[1];\r
-- bufferBytes *= formatBytes( stream_.userFormat );\r
-- }\r
--\r
-- LPDIRECTSOUNDCAPTUREBUFFER dsBuffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1];\r
-- long nextReadPointer = handle->bufferPointer[1];\r
-- DWORD dsBufferSize = handle->dsBufferSize[1];\r
--\r
-- // Find out where the write and "safe read" pointers are.\r
-- result = dsBuffer->GetCurrentPosition( ¤tReadPointer, &safeReadPointer );\r
-- if ( FAILED( result ) ) {\r
-- errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current read position!";\r
-- errorText_ = errorStream_.str();\r
-- MUTEX_UNLOCK( &stream_.mutex );\r
-- error( RtAudioError::SYSTEM_ERROR );\r
-- return;\r
-- }\r
--\r
-- if ( safeReadPointer < (DWORD)nextReadPointer ) safeReadPointer += dsBufferSize; // unwrap offset\r
-- DWORD endRead = nextReadPointer + bufferBytes;\r
--\r
-- // Handling depends on whether we are INPUT or DUPLEX.\r
-- // If we're in INPUT mode then waiting is a good thing. If we're in DUPLEX mode,\r
-- // then a wait here will drag the write pointers into the forbidden zone.\r
-- //\r
-- // In DUPLEX mode, rather than wait, we will back off the read pointer until\r
-- // it's in a safe position. This causes dropouts, but it seems to be the only\r
-- // practical way to sync up the read and write pointers reliably, given the\r
-- // the very complex relationship between phase and increment of the read and write\r
-- // pointers.\r
-- //\r
-- // In order to minimize audible dropouts in DUPLEX mode, we will\r
-- // provide a pre-roll period of 0.5 seconds in which we return\r
-- // zeros from the read buffer while the pointers sync up.\r
--\r
-- if ( stream_.mode == DUPLEX ) {\r
-- if ( safeReadPointer < endRead ) {\r
-- if ( duplexPrerollBytes <= 0 ) {\r
-- // Pre-roll time over. Be more agressive.\r
-- int adjustment = endRead-safeReadPointer;\r
--\r
-- handle->xrun[1] = true;\r
-- // Two cases:\r
-- // - large adjustments: we've probably run out of CPU cycles, so just resync exactly,\r
-- // and perform fine adjustments later.\r
-- // - small adjustments: back off by twice as much.\r
-- if ( adjustment >= 2*bufferBytes )\r
-- nextReadPointer = safeReadPointer-2*bufferBytes;\r
-- else\r
-- nextReadPointer = safeReadPointer-bufferBytes-adjustment;\r
--\r
-- if ( nextReadPointer < 0 ) nextReadPointer += dsBufferSize;\r
--\r
-- }\r
-- else {\r
-- // In pre=roll time. Just do it.\r
-- nextReadPointer = safeReadPointer - bufferBytes;\r
-- while ( nextReadPointer < 0 ) nextReadPointer += dsBufferSize;\r
-- }\r
-- endRead = nextReadPointer + bufferBytes;\r
-- }\r
-- }\r
-- else { // mode == INPUT\r
-- while ( safeReadPointer < endRead && stream_.callbackInfo.isRunning ) {\r
-- // See comments for playback.\r
-- double millis = (endRead - safeReadPointer) * 1000.0;\r
-- millis /= ( formatBytes(stream_.deviceFormat[1]) * stream_.nDeviceChannels[1] * stream_.sampleRate);\r
-- if ( millis < 1.0 ) millis = 1.0;\r
-- Sleep( (DWORD) millis );\r
--\r
-- // Wake up and find out where we are now.\r
-- result = dsBuffer->GetCurrentPosition( ¤tReadPointer, &safeReadPointer );\r
-- if ( FAILED( result ) ) {\r
-- errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current read position!";\r
-- errorText_ = errorStream_.str();\r
-- MUTEX_UNLOCK( &stream_.mutex );\r
-- error( RtAudioError::SYSTEM_ERROR );\r
-- return;\r
-- }\r
--\r
-- if ( safeReadPointer < (DWORD)nextReadPointer ) safeReadPointer += dsBufferSize; // unwrap offset\r
-- }\r
-- }\r
--\r
-- // Lock free space in the buffer\r
-- result = dsBuffer->Lock( nextReadPointer, bufferBytes, &buffer1,\r
-- &bufferSize1, &buffer2, &bufferSize2, 0 );\r
-- if ( FAILED( result ) ) {\r
-- errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") locking capture buffer!";\r
-- errorText_ = errorStream_.str();\r
-- MUTEX_UNLOCK( &stream_.mutex );\r
-- error( RtAudioError::SYSTEM_ERROR );\r
-- return;\r
-- }\r
--\r
-- if ( duplexPrerollBytes <= 0 ) {\r
-- // Copy our buffer into the DS buffer\r
-- CopyMemory( buffer, buffer1, bufferSize1 );\r
-- if ( buffer2 != NULL ) CopyMemory( buffer+bufferSize1, buffer2, bufferSize2 );\r
-- }\r
-- else {\r
-- memset( buffer, 0, bufferSize1 );\r
-- if ( buffer2 != NULL ) memset( buffer + bufferSize1, 0, bufferSize2 );\r
-- duplexPrerollBytes -= bufferSize1 + bufferSize2;\r
-- }\r
--\r
-- // Update our buffer offset and unlock sound buffer\r
-- nextReadPointer = ( nextReadPointer + bufferSize1 + bufferSize2 ) % dsBufferSize;\r
-- dsBuffer->Unlock( buffer1, bufferSize1, buffer2, bufferSize2 );\r
-- if ( FAILED( result ) ) {\r
-- errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") unlocking capture buffer!";\r
-- errorText_ = errorStream_.str();\r
-- MUTEX_UNLOCK( &stream_.mutex );\r
-- error( RtAudioError::SYSTEM_ERROR );\r
-- return;\r
-- }\r
-- handle->bufferPointer[1] = nextReadPointer;\r
--\r
-- // No byte swapping necessary in DirectSound implementation.\r
--\r
-- // If necessary, convert 8-bit data from unsigned to signed.\r
-- if ( stream_.deviceFormat[1] == RTAUDIO_SINT8 )\r
-- for ( int j=0; j<bufferBytes; j++ ) buffer[j] = (signed char) ( buffer[j] - 128 );\r
--\r
-- // Do buffer conversion if necessary.\r
-- if ( stream_.doConvertBuffer[1] )\r
-- convertBuffer( stream_.userBuffer[1], stream_.deviceBuffer, stream_.convertInfo[1] );\r
-- }\r
--\r
-- unlock:\r
-- MUTEX_UNLOCK( &stream_.mutex );\r
-- RtApi::tickStreamTime();\r
--}\r
--\r
--// Definitions for utility functions and callbacks\r
--// specific to the DirectSound implementation.\r
--\r
--static unsigned __stdcall callbackHandler( void *ptr )\r
--{\r
-- CallbackInfo *info = (CallbackInfo *) ptr;\r
-- RtApiDs *object = (RtApiDs *) info->object;\r
-- bool* isRunning = &info->isRunning;\r
--\r
-- while ( *isRunning == true ) {\r
-- object->callbackEvent();\r
-- }\r
--\r
-- _endthreadex( 0 );\r
-- return 0;\r
--}\r
--\r
--static BOOL CALLBACK deviceQueryCallback( LPGUID lpguid,\r
-- LPCTSTR description,\r
-- LPCTSTR /*module*/,\r
-- LPVOID lpContext )\r
--{\r
-- struct DsProbeData& probeInfo = *(struct DsProbeData*) lpContext;\r
-- std::vector<struct DsDevice>& dsDevices = *probeInfo.dsDevices;\r
--\r
-- HRESULT hr;\r
-- bool validDevice = false;\r
-- if ( probeInfo.isInput == true ) {\r
-- DSCCAPS caps;\r
-- LPDIRECTSOUNDCAPTURE object;\r
--\r
-- hr = DirectSoundCaptureCreate( lpguid, &object, NULL );\r
-- if ( hr != DS_OK ) return TRUE;\r
--\r
-- caps.dwSize = sizeof(caps);\r
-- hr = object->GetCaps( &caps );\r
-- if ( hr == DS_OK ) {\r
-- if ( caps.dwChannels > 0 && caps.dwFormats > 0 )\r
-- validDevice = true;\r
-- }\r
-- object->Release();\r
-- }\r
-- else {\r
-- DSCAPS caps;\r
-- LPDIRECTSOUND object;\r
-- hr = DirectSoundCreate( lpguid, &object, NULL );\r
-- if ( hr != DS_OK ) return TRUE;\r
--\r
-- caps.dwSize = sizeof(caps);\r
-- hr = object->GetCaps( &caps );\r
-- if ( hr == DS_OK ) {\r
-- if ( caps.dwFlags & DSCAPS_PRIMARYMONO || caps.dwFlags & DSCAPS_PRIMARYSTEREO )\r
-- validDevice = true;\r
-- }\r
-- object->Release();\r
-- }\r
--\r
-- // If good device, then save its name and guid.\r
-- std::string name = convertCharPointerToStdString( description );\r
-- //if ( name == "Primary Sound Driver" || name == "Primary Sound Capture Driver" )\r
-- if ( lpguid == NULL )\r
-- name = "Default Device";\r
-- if ( validDevice ) {\r
-- for ( unsigned int i=0; i<dsDevices.size(); i++ ) {\r
-- if ( dsDevices[i].name == name ) {\r
-- dsDevices[i].found = true;\r
-- if ( probeInfo.isInput ) {\r
-- dsDevices[i].id[1] = lpguid;\r
-- dsDevices[i].validId[1] = true;\r
-- }\r
-- else {\r
-- dsDevices[i].id[0] = lpguid;\r
-- dsDevices[i].validId[0] = true;\r
-- }\r
-- return TRUE;\r
-- }\r
-- }\r
--\r
-- DsDevice device;\r
-- device.name = name;\r
-- device.found = true;\r
-- if ( probeInfo.isInput ) {\r
-- device.id[1] = lpguid;\r
-- device.validId[1] = true;\r
-- }\r
-- else {\r
-- device.id[0] = lpguid;\r
-- device.validId[0] = true;\r
-- }\r
-- dsDevices.push_back( device );\r
-- }\r
--\r
-- return TRUE;\r
--}\r
--\r
--static const char* getErrorString( int code )\r
--{\r
-- switch ( code ) {\r
--\r
-- case DSERR_ALLOCATED:\r
-- return "Already allocated";\r
--\r
-- case DSERR_CONTROLUNAVAIL:\r
-- return "Control unavailable";\r
--\r
-- case DSERR_INVALIDPARAM:\r
-- return "Invalid parameter";\r
--\r
-- case DSERR_INVALIDCALL:\r
-- return "Invalid call";\r
--\r
-- case DSERR_GENERIC:\r
-- return "Generic error";\r
--\r
-- case DSERR_PRIOLEVELNEEDED:\r
-- return "Priority level needed";\r
--\r
-- case DSERR_OUTOFMEMORY:\r
-- return "Out of memory";\r
--\r
-- case DSERR_BADFORMAT:\r
-- return "The sample rate or the channel format is not supported";\r
--\r
-- case DSERR_UNSUPPORTED:\r
-- return "Not supported";\r
--\r
-- case DSERR_NODRIVER:\r
-- return "No driver";\r
--\r
-- case DSERR_ALREADYINITIALIZED:\r
-- return "Already initialized";\r
--\r
-- case DSERR_NOAGGREGATION:\r
-- return "No aggregation";\r
--\r
-- case DSERR_BUFFERLOST:\r
-- return "Buffer lost";\r
--\r
-- case DSERR_OTHERAPPHASPRIO:\r
-- return "Another application already has priority";\r
--\r
-- case DSERR_UNINITIALIZED:\r
-- return "Uninitialized";\r
--\r
-- default:\r
-- return "DirectSound unknown error";\r
-- }\r
--}\r
--//******************** End of __WINDOWS_DS__ *********************//\r
--#endif\r
--\r
--\r
--#if defined(__LINUX_ALSA__)\r
--\r
--#include <alsa/asoundlib.h>\r
--#include <unistd.h>\r
--\r
-- // A structure to hold various information related to the ALSA API\r
-- // implementation.\r
--struct AlsaHandle {\r
-- snd_pcm_t *handles[2];\r
-- bool synchronized;\r
-- bool xrun[2];\r
-- pthread_cond_t runnable_cv;\r
-- bool runnable;\r
--\r
-- AlsaHandle()\r
-- :synchronized(false), runnable(false) { xrun[0] = false; xrun[1] = false; }\r
--};\r
--\r
--static void *alsaCallbackHandler( void * ptr );\r
--\r
--RtApiAlsa :: RtApiAlsa()\r
--{\r
-- // Nothing to do here.\r
--}\r
--\r
--RtApiAlsa :: ~RtApiAlsa()\r
--{\r
-- if ( stream_.state != STREAM_CLOSED ) closeStream();\r
--}\r
--\r
--unsigned int RtApiAlsa :: getDeviceCount( void )\r
--{\r
-- unsigned nDevices = 0;\r
-- int result, subdevice, card;\r
-- char name[64];\r
-- snd_ctl_t *handle;\r
--\r
-- // Count cards and devices\r
-- card = -1;\r
-- snd_card_next( &card );\r
-- while ( card >= 0 ) {\r
-- sprintf( name, "hw:%d", card );\r
-- result = snd_ctl_open( &handle, name, 0 );\r
-- if ( result < 0 ) {\r
-- errorStream_ << "RtApiAlsa::getDeviceCount: control open, card = " << card << ", " << snd_strerror( result ) << ".";\r
-- errorText_ = errorStream_.str();\r
-- error( RtAudioError::WARNING );\r
-- goto nextcard;\r
-- }\r
-- subdevice = -1;\r
-- while( 1 ) {\r
-- result = snd_ctl_pcm_next_device( handle, &subdevice );\r
-- if ( result < 0 ) {\r
-- errorStream_ << "RtApiAlsa::getDeviceCount: control next device, card = " << card << ", " << snd_strerror( result ) << ".";\r
-- errorText_ = errorStream_.str();\r
-- error( RtAudioError::WARNING );\r
-- break;\r
-- }\r
-- if ( subdevice < 0 )\r
-- break;\r
-- nDevices++;\r
-- }\r
-- nextcard:\r
-- snd_ctl_close( handle );\r
-- snd_card_next( &card );\r
-- }\r
--\r
-- result = snd_ctl_open( &handle, "default", 0 );\r
-- if (result == 0) {\r
-- nDevices++;\r
-- snd_ctl_close( handle );\r
-- }\r
--\r
-- return nDevices;\r
--}\r
--\r
--RtAudio::DeviceInfo RtApiAlsa :: getDeviceInfo( unsigned int device )\r
--{\r
-- RtAudio::DeviceInfo info;\r
-- info.probed = false;\r
--\r
-- unsigned nDevices = 0;\r
-- int result, subdevice, card;\r
-- char name[64];\r
-- snd_ctl_t *chandle;\r
--\r
-- // Count cards and devices\r
-- card = -1;\r
-- subdevice = -1;\r
-- snd_card_next( &card );\r
-- while ( card >= 0 ) {\r
-- sprintf( name, "hw:%d", card );\r
-- result = snd_ctl_open( &chandle, name, SND_CTL_NONBLOCK );\r
-- if ( result < 0 ) {\r
-- errorStream_ << "RtApiAlsa::getDeviceInfo: control open, card = " << card << ", " << snd_strerror( result ) << ".";\r
-- errorText_ = errorStream_.str();\r
-- error( RtAudioError::WARNING );\r
-- goto nextcard;\r
-- }\r
-- subdevice = -1;\r
-- while( 1 ) {\r
-- result = snd_ctl_pcm_next_device( chandle, &subdevice );\r
-- if ( result < 0 ) {\r
-- errorStream_ << "RtApiAlsa::getDeviceInfo: control next device, card = " << card << ", " << snd_strerror( result ) << ".";\r
-- errorText_ = errorStream_.str();\r
-- error( RtAudioError::WARNING );\r
-- break;\r
-- }\r
-- if ( subdevice < 0 ) break;\r
-- if ( nDevices == device ) {\r
-- sprintf( name, "hw:%d,%d", card, subdevice );\r
-- goto foundDevice;\r
-- }\r
-- nDevices++;\r
-- }\r
-- nextcard:\r
-- snd_ctl_close( chandle );\r
-- snd_card_next( &card );\r
-- }\r
--\r
-- result = snd_ctl_open( &chandle, "default", SND_CTL_NONBLOCK );\r
-- if ( result == 0 ) {\r
-- if ( nDevices == device ) {\r
-- strcpy( name, "default" );\r
-- goto foundDevice;\r
-- }\r
-- nDevices++;\r
-- }\r
--\r
-- if ( nDevices == 0 ) {\r
-- errorText_ = "RtApiAlsa::getDeviceInfo: no devices found!";\r
-- error( RtAudioError::INVALID_USE );\r
-- return info;\r
-- }\r
--\r
-- if ( device >= nDevices ) {\r
-- errorText_ = "RtApiAlsa::getDeviceInfo: device ID is invalid!";\r
-- error( RtAudioError::INVALID_USE );\r
-- return info;\r
-- }\r
--\r
-- foundDevice:\r
--\r
-- // If a stream is already open, we cannot probe the stream devices.\r
-- // Thus, use the saved results.\r
-- if ( stream_.state != STREAM_CLOSED &&\r
-- ( stream_.device[0] == device || stream_.device[1] == device ) ) {\r
-- snd_ctl_close( chandle );\r
-- if ( device >= devices_.size() ) {\r
-- errorText_ = "RtApiAlsa::getDeviceInfo: device ID was not present before stream was opened.";\r
-- error( RtAudioError::WARNING );\r
-- return info;\r
-- }\r
-- return devices_[ device ];\r
-- }\r
--\r
-- int openMode = SND_PCM_ASYNC;\r
-- snd_pcm_stream_t stream;\r
-- snd_pcm_info_t *pcminfo;\r
-- snd_pcm_info_alloca( &pcminfo );\r
-- snd_pcm_t *phandle;\r
-- snd_pcm_hw_params_t *params;\r
-- snd_pcm_hw_params_alloca( ¶ms );\r
--\r
-- // First try for playback unless default device (which has subdev -1)\r
-- stream = SND_PCM_STREAM_PLAYBACK;\r
-- snd_pcm_info_set_stream( pcminfo, stream );\r
-- if ( subdevice != -1 ) {\r
-- snd_pcm_info_set_device( pcminfo, subdevice );\r
-- snd_pcm_info_set_subdevice( pcminfo, 0 );\r
--\r
-- result = snd_ctl_pcm_info( chandle, pcminfo );\r
-- if ( result < 0 ) {\r
-- // Device probably doesn't support playback.\r
-- goto captureProbe;\r
-- }\r
-- }\r
--\r
-- result = snd_pcm_open( &phandle, name, stream, openMode | SND_PCM_NONBLOCK );\r
-- if ( result < 0 ) {\r
-- errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_open error for device (" << name << "), " << snd_strerror( result ) << ".";\r
-- errorText_ = errorStream_.str();\r
-- error( RtAudioError::WARNING );\r
-- goto captureProbe;\r
-- }\r
--\r
-- // The device is open ... fill the parameter structure.\r
-- result = snd_pcm_hw_params_any( phandle, params );\r
-- if ( result < 0 ) {\r
-- snd_pcm_close( phandle );\r
-- errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_hw_params error for device (" << name << "), " << snd_strerror( result ) << ".";\r
-- errorText_ = errorStream_.str();\r
-- error( RtAudioError::WARNING );\r
-- goto captureProbe;\r
-- }\r
--\r
-- // Get output channel information.\r
-- unsigned int value;\r
-- result = snd_pcm_hw_params_get_channels_max( params, &value );\r
-- if ( result < 0 ) {\r
-- snd_pcm_close( phandle );\r
-- errorStream_ << "RtApiAlsa::getDeviceInfo: error getting device (" << name << ") output channels, " << snd_strerror( result ) << ".";\r
-- errorText_ = errorStream_.str();\r
-- error( RtAudioError::WARNING );\r
-- goto captureProbe;\r
-- }\r
-- info.outputChannels = value;\r
-- snd_pcm_close( phandle );\r
--\r
-- captureProbe:\r
-- stream = SND_PCM_STREAM_CAPTURE;\r
-- snd_pcm_info_set_stream( pcminfo, stream );\r
--\r
-- // Now try for capture unless default device (with subdev = -1)\r
-- if ( subdevice != -1 ) {\r
-- result = snd_ctl_pcm_info( chandle, pcminfo );\r
-- snd_ctl_close( chandle );\r
-- if ( result < 0 ) {\r
-- // Device probably doesn't support capture.\r
-- if ( info.outputChannels == 0 ) return info;\r
-- goto probeParameters;\r
-- }\r
-- }\r
-- else\r
-- snd_ctl_close( chandle );\r
--\r
-- result = snd_pcm_open( &phandle, name, stream, openMode | SND_PCM_NONBLOCK);\r
-- if ( result < 0 ) {\r
-- errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_open error for device (" << name << "), " << snd_strerror( result ) << ".";\r
-- errorText_ = errorStream_.str();\r
-- error( RtAudioError::WARNING );\r
-- if ( info.outputChannels == 0 ) return info;\r
-- goto probeParameters;\r
-- }\r
--\r
-- // The device is open ... fill the parameter structure.\r
-- result = snd_pcm_hw_params_any( phandle, params );\r
-- if ( result < 0 ) {\r
-- snd_pcm_close( phandle );\r
-- errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_hw_params error for device (" << name << "), " << snd_strerror( result ) << ".";\r
-- errorText_ = errorStream_.str();\r
-- error( RtAudioError::WARNING );\r
-- if ( info.outputChannels == 0 ) return info;\r
-- goto probeParameters;\r
-- }\r
--\r
-- result = snd_pcm_hw_params_get_channels_max( params, &value );\r
-- if ( result < 0 ) {\r
-- snd_pcm_close( phandle );\r
-- errorStream_ << "RtApiAlsa::getDeviceInfo: error getting device (" << name << ") input channels, " << snd_strerror( result ) << ".";\r
-- errorText_ = errorStream_.str();\r
-- error( RtAudioError::WARNING );\r
-- if ( info.outputChannels == 0 ) return info;\r
-- goto probeParameters;\r
-- }\r
-- info.inputChannels = value;\r
-- snd_pcm_close( phandle );\r
--\r
-- // If device opens for both playback and capture, we determine the channels.\r
-- if ( info.outputChannels > 0 && info.inputChannels > 0 )\r
-- info.duplexChannels = (info.outputChannels > info.inputChannels) ? info.inputChannels : info.outputChannels;\r
--\r
-- // ALSA doesn't provide default devices so we'll use the first available one.\r
-- if ( device == 0 && info.outputChannels > 0 )\r
-- info.isDefaultOutput = true;\r
-- if ( device == 0 && info.inputChannels > 0 )\r
-- info.isDefaultInput = true;\r
--\r
-- probeParameters:\r
-- // At this point, we just need to figure out the supported data\r
-- // formats and sample rates. We'll proceed by opening the device in\r
-- // the direction with the maximum number of channels, or playback if\r
-- // they are equal. This might limit our sample rate options, but so\r
-- // be it.\r
--\r
-- if ( info.outputChannels >= info.inputChannels )\r
-- stream = SND_PCM_STREAM_PLAYBACK;\r
-- else\r
-- stream = SND_PCM_STREAM_CAPTURE;\r
-- snd_pcm_info_set_stream( pcminfo, stream );\r
--\r
-- result = snd_pcm_open( &phandle, name, stream, openMode | SND_PCM_NONBLOCK);\r
-- if ( result < 0 ) {\r
-- errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_open error for device (" << name << "), " << snd_strerror( result ) << ".";\r
-- errorText_ = errorStream_.str();\r
-- error( RtAudioError::WARNING );\r
-- return info;\r
-- }\r
--\r
-- // The device is open ... fill the parameter structure.\r
-- result = snd_pcm_hw_params_any( phandle, params );\r
-- if ( result < 0 ) {\r
-- snd_pcm_close( phandle );\r
-- errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_hw_params error for device (" << name << "), " << snd_strerror( result ) << ".";\r
-- errorText_ = errorStream_.str();\r
-- error( RtAudioError::WARNING );\r
-- return info;\r
-- }\r
--\r
-- // Test our discrete set of sample rate values.\r
-- info.sampleRates.clear();\r
-- for ( unsigned int i=0; i<MAX_SAMPLE_RATES; i++ ) {\r
-- if ( snd_pcm_hw_params_test_rate( phandle, params, SAMPLE_RATES[i], 0 ) == 0 ) {\r
-- info.sampleRates.push_back( SAMPLE_RATES[i] );\r
--\r
-- if ( !info.preferredSampleRate || ( SAMPLE_RATES[i] <= 48000 && SAMPLE_RATES[i] > info.preferredSampleRate ) )\r
-- info.preferredSampleRate = SAMPLE_RATES[i];\r
-- }\r
-- }\r
-- if ( info.sampleRates.size() == 0 ) {\r
-- snd_pcm_close( phandle );\r
-- errorStream_ << "RtApiAlsa::getDeviceInfo: no supported sample rates found for device (" << name << ").";\r
-- errorText_ = errorStream_.str();\r
-- error( RtAudioError::WARNING );\r
-- return info;\r
-- }\r
--\r
-- // Probe the supported data formats ... we don't care about endian-ness just yet\r
-- snd_pcm_format_t format;\r
-- info.nativeFormats = 0;\r
-- format = SND_PCM_FORMAT_S8;\r
-- if ( snd_pcm_hw_params_test_format( phandle, params, format ) == 0 )\r
-- info.nativeFormats |= RTAUDIO_SINT8;\r
-- format = SND_PCM_FORMAT_S16;\r
-- if ( snd_pcm_hw_params_test_format( phandle, params, format ) == 0 )\r
-- info.nativeFormats |= RTAUDIO_SINT16;\r
-- format = SND_PCM_FORMAT_S24;\r
-- if ( snd_pcm_hw_params_test_format( phandle, params, format ) == 0 )\r
-- info.nativeFormats |= RTAUDIO_SINT24;\r
-- format = SND_PCM_FORMAT_S32;\r
-- if ( snd_pcm_hw_params_test_format( phandle, params, format ) == 0 )\r
-- info.nativeFormats |= RTAUDIO_SINT32;\r
-- format = SND_PCM_FORMAT_FLOAT;\r
-- if ( snd_pcm_hw_params_test_format( phandle, params, format ) == 0 )\r
-- info.nativeFormats |= RTAUDIO_FLOAT32;\r
-- format = SND_PCM_FORMAT_FLOAT64;\r
-- if ( snd_pcm_hw_params_test_format( phandle, params, format ) == 0 )\r
-- info.nativeFormats |= RTAUDIO_FLOAT64;\r
--\r
-- // Check that we have at least one supported format\r
-- if ( info.nativeFormats == 0 ) {\r
-- snd_pcm_close( phandle );\r
-- errorStream_ << "RtApiAlsa::getDeviceInfo: pcm device (" << name << ") data format not supported by RtAudio.";\r
-- errorText_ = errorStream_.str();\r
-- error( RtAudioError::WARNING );\r
-- return info;\r
-- }\r
--\r
-- // Get the device name\r
-- char *cardname;\r
-- result = snd_card_get_name( card, &cardname );\r
-- if ( result >= 0 ) {\r
-- sprintf( name, "hw:%s,%d", cardname, subdevice );\r
-- free( cardname );\r
-- }\r
-- info.name = name;\r
--\r
-- // That's all ... close the device and return\r
-- snd_pcm_close( phandle );\r
-- info.probed = true;\r
-- return info;\r
--}\r
--\r
--void RtApiAlsa :: saveDeviceInfo( void )\r
--{\r
-- devices_.clear();\r
--\r
-- unsigned int nDevices = getDeviceCount();\r
-- devices_.resize( nDevices );\r
-- for ( unsigned int i=0; i<nDevices; i++ )\r
-- devices_[i] = getDeviceInfo( i );\r
--}\r
--\r
--bool RtApiAlsa :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,\r
-- unsigned int firstChannel, unsigned int sampleRate,\r
-- RtAudioFormat format, unsigned int *bufferSize,\r
-- RtAudio::StreamOptions *options )\r
--\r
--{\r
--#if defined(__RTAUDIO_DEBUG__)\r
-- snd_output_t *out;\r
-- snd_output_stdio_attach(&out, stderr, 0);\r
--#endif\r
--\r
-- // I'm not using the "plug" interface ... too much inconsistent behavior.\r
--\r
-- unsigned nDevices = 0;\r
-- int result, subdevice, card;\r
-- char name[64];\r
-- snd_ctl_t *chandle;\r
--\r
-- if ( options && options->flags & RTAUDIO_ALSA_USE_DEFAULT )\r
-- snprintf(name, sizeof(name), "%s", "default");\r
-- else {\r
-- // Count cards and devices\r
-- card = -1;\r
-- snd_card_next( &card );\r
-- while ( card >= 0 ) {\r
-- sprintf( name, "hw:%d", card );\r
-- result = snd_ctl_open( &chandle, name, SND_CTL_NONBLOCK );\r
-- if ( result < 0 ) {\r
-- errorStream_ << "RtApiAlsa::probeDeviceOpen: control open, card = " << card << ", " << snd_strerror( result ) << ".";\r
-- errorText_ = errorStream_.str();\r
-- return FAILURE;\r
-- }\r
-- subdevice = -1;\r
-- while( 1 ) {\r
-- result = snd_ctl_pcm_next_device( chandle, &subdevice );\r
-- if ( result < 0 ) break;\r
-- if ( subdevice < 0 ) break;\r
-- if ( nDevices == device ) {\r
-- sprintf( name, "hw:%d,%d", card, subdevice );\r
-- snd_ctl_close( chandle );\r
-- goto foundDevice;\r
-- }\r
-- nDevices++;\r
-- }\r
-- snd_ctl_close( chandle );\r
-- snd_card_next( &card );\r
-- }\r
--\r
-- result = snd_ctl_open( &chandle, "default", SND_CTL_NONBLOCK );\r
-- if ( result == 0 ) {\r
-- if ( nDevices == device ) {\r
-- strcpy( name, "default" );\r
-- goto foundDevice;\r
-- }\r
-- nDevices++;\r
-- }\r
--\r
-- if ( nDevices == 0 ) {\r
-- // This should not happen because a check is made before this function is called.\r
-- errorText_ = "RtApiAlsa::probeDeviceOpen: no devices found!";\r
-- return FAILURE;\r
-- }\r
--\r
-- if ( device >= nDevices ) {\r
-- // This should not happen because a check is made before this function is called.\r
-- errorText_ = "RtApiAlsa::probeDeviceOpen: device ID is invalid!";\r
-- return FAILURE;\r
-- }\r
-- }\r
--\r
-- foundDevice:\r
--\r
-- // The getDeviceInfo() function will not work for a device that is\r
-- // already open. Thus, we'll probe the system before opening a\r
-- // stream and save the results for use by getDeviceInfo().\r
-- if ( mode == OUTPUT || ( mode == INPUT && stream_.mode != OUTPUT ) ) // only do once\r
-- this->saveDeviceInfo();\r
--\r
-- snd_pcm_stream_t stream;\r
-- if ( mode == OUTPUT )\r
-- stream = SND_PCM_STREAM_PLAYBACK;\r
-- else\r
-- stream = SND_PCM_STREAM_CAPTURE;\r
--\r
-- snd_pcm_t *phandle;\r
-- int openMode = SND_PCM_ASYNC;\r
-- result = snd_pcm_open( &phandle, name, stream, openMode );\r
-- if ( result < 0 ) {\r
-- if ( mode == OUTPUT )\r
-- errorStream_ << "RtApiAlsa::probeDeviceOpen: pcm device (" << name << ") won't open for output.";\r
-- else\r
-- errorStream_ << "RtApiAlsa::probeDeviceOpen: pcm device (" << name << ") won't open for input.";\r
-- errorText_ = errorStream_.str();\r
-- return FAILURE;\r
-- }\r
--\r
-- // Fill the parameter structure.\r
-- snd_pcm_hw_params_t *hw_params;\r
-- snd_pcm_hw_params_alloca( &hw_params );\r
-- result = snd_pcm_hw_params_any( phandle, hw_params );\r
-- if ( result < 0 ) {\r
-- snd_pcm_close( phandle );\r
-- errorStream_ << "RtApiAlsa::probeDeviceOpen: error getting pcm device (" << name << ") parameters, " << snd_strerror( result ) << ".";\r
-- errorText_ = errorStream_.str();\r
-- return FAILURE;\r
-- }\r
--\r
--#if defined(__RTAUDIO_DEBUG__)\r
-- fprintf( stderr, "\nRtApiAlsa: dump hardware params just after device open:\n\n" );\r
-- snd_pcm_hw_params_dump( hw_params, out );\r
--#endif\r
--\r
-- // Set access ... check user preference.\r
-- if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) {\r
-- stream_.userInterleaved = false;\r
-- result = snd_pcm_hw_params_set_access( phandle, hw_params, SND_PCM_ACCESS_RW_NONINTERLEAVED );\r
-- if ( result < 0 ) {\r
-- result = snd_pcm_hw_params_set_access( phandle, hw_params, SND_PCM_ACCESS_RW_INTERLEAVED );\r
-- stream_.deviceInterleaved[mode] = true;\r
-- }\r
-- else\r
-- stream_.deviceInterleaved[mode] = false;\r
-- }\r
-- else {\r
-- stream_.userInterleaved = true;\r
-- result = snd_pcm_hw_params_set_access( phandle, hw_params, SND_PCM_ACCESS_RW_INTERLEAVED );\r
-- if ( result < 0 ) {\r
-- result = snd_pcm_hw_params_set_access( phandle, hw_params, SND_PCM_ACCESS_RW_NONINTERLEAVED );\r
-- stream_.deviceInterleaved[mode] = false;\r
-- }\r
-- else\r
-- stream_.deviceInterleaved[mode] = true;\r
-- }\r
--\r
-- if ( result < 0 ) {\r
-- snd_pcm_close( phandle );\r
-- errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting pcm device (" << name << ") access, " << snd_strerror( result ) << ".";\r
-- errorText_ = errorStream_.str();\r
-- return FAILURE;\r
-- }\r
--\r
-- // Determine how to set the device format.\r
-- stream_.userFormat = format;\r
-- snd_pcm_format_t deviceFormat = SND_PCM_FORMAT_UNKNOWN;\r
--\r
-- if ( format == RTAUDIO_SINT8 )\r
-- deviceFormat = SND_PCM_FORMAT_S8;\r
-- else if ( format == RTAUDIO_SINT16 )\r
-- deviceFormat = SND_PCM_FORMAT_S16;\r
-- else if ( format == RTAUDIO_SINT24 )\r
-- deviceFormat = SND_PCM_FORMAT_S24;\r
-- else if ( format == RTAUDIO_SINT32 )\r
-- deviceFormat = SND_PCM_FORMAT_S32;\r
-- else if ( format == RTAUDIO_FLOAT32 )\r
-- deviceFormat = SND_PCM_FORMAT_FLOAT;\r
-- else if ( format == RTAUDIO_FLOAT64 )\r
-- deviceFormat = SND_PCM_FORMAT_FLOAT64;\r
--\r
-- if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat) == 0) {\r
-- stream_.deviceFormat[mode] = format;\r
-- goto setFormat;\r
-- }\r
--\r
-- // The user requested format is not natively supported by the device.\r
-- deviceFormat = SND_PCM_FORMAT_FLOAT64;\r
-- if ( snd_pcm_hw_params_test_format( phandle, hw_params, deviceFormat ) == 0 ) {\r
-- stream_.deviceFormat[mode] = RTAUDIO_FLOAT64;\r
-- goto setFormat;\r
-- }\r
--\r
-- deviceFormat = SND_PCM_FORMAT_FLOAT;\r
-- if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat ) == 0 ) {\r
-- stream_.deviceFormat[mode] = RTAUDIO_FLOAT32;\r
-- goto setFormat;\r
-- }\r
--\r
-- deviceFormat = SND_PCM_FORMAT_S32;\r
-- if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat ) == 0 ) {\r
-- stream_.deviceFormat[mode] = RTAUDIO_SINT32;\r
-- goto setFormat;\r
-- }\r
--\r
-- deviceFormat = SND_PCM_FORMAT_S24;\r
-- if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat ) == 0 ) {\r
-- stream_.deviceFormat[mode] = RTAUDIO_SINT24;\r
-- goto setFormat;\r
-- }\r
--\r
-- deviceFormat = SND_PCM_FORMAT_S16;\r
-- if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat ) == 0 ) {\r
-- stream_.deviceFormat[mode] = RTAUDIO_SINT16;\r
-- goto setFormat;\r
-- }\r
--\r
-- deviceFormat = SND_PCM_FORMAT_S8;\r
-- if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat ) == 0 ) {\r
-- stream_.deviceFormat[mode] = RTAUDIO_SINT8;\r
-- goto setFormat;\r
-- }\r
--\r
-- // If we get here, no supported format was found.\r
-- snd_pcm_close( phandle );\r
-- errorStream_ << "RtApiAlsa::probeDeviceOpen: pcm device " << device << " data format not supported by RtAudio.";\r
-- errorText_ = errorStream_.str();\r
-- return FAILURE;\r
--\r
-- setFormat:\r
-- result = snd_pcm_hw_params_set_format( phandle, hw_params, deviceFormat );\r
-- if ( result < 0 ) {\r
-- snd_pcm_close( phandle );\r
-- errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting pcm device (" << name << ") data format, " << snd_strerror( result ) << ".";\r
-- errorText_ = errorStream_.str();\r
-- return FAILURE;\r
-- }\r
--\r
-- // Determine whether byte-swaping is necessary.\r
-- stream_.doByteSwap[mode] = false;\r
-- if ( deviceFormat != SND_PCM_FORMAT_S8 ) {\r
-- result = snd_pcm_format_cpu_endian( deviceFormat );\r
-- if ( result == 0 )\r
-- stream_.doByteSwap[mode] = true;\r
-- else if (result < 0) {\r
-- snd_pcm_close( phandle );\r
-- errorStream_ << "RtApiAlsa::probeDeviceOpen: error getting pcm device (" << name << ") endian-ness, " << snd_strerror( result ) << ".";\r
-- errorText_ = errorStream_.str();\r
-- return FAILURE;\r
-- }\r
-- }\r
--\r
-- // Set the sample rate.\r
-- result = snd_pcm_hw_params_set_rate_near( phandle, hw_params, (unsigned int*) &sampleRate, 0 );\r
-- if ( result < 0 ) {\r
-- snd_pcm_close( phandle );\r
-- errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting sample rate on device (" << name << "), " << snd_strerror( result ) << ".";\r
-- errorText_ = errorStream_.str();\r
-- return FAILURE;\r
-- }\r
--\r
-- // Determine the number of channels for this device. We support a possible\r
-- // minimum device channel number > than the value requested by the user.\r
-- stream_.nUserChannels[mode] = channels;\r
-- unsigned int value;\r
-- result = snd_pcm_hw_params_get_channels_max( hw_params, &value );\r
-- unsigned int deviceChannels = value;\r
-- if ( result < 0 || deviceChannels < channels + firstChannel ) {\r
-- snd_pcm_close( phandle );\r
-- errorStream_ << "RtApiAlsa::probeDeviceOpen: requested channel parameters not supported by device (" << name << "), " << snd_strerror( result ) << ".";\r
-- errorText_ = errorStream_.str();\r
-- return FAILURE;\r
-- }\r
--\r
-- result = snd_pcm_hw_params_get_channels_min( hw_params, &value );\r
-- if ( result < 0 ) {\r
-- snd_pcm_close( phandle );\r
-- errorStream_ << "RtApiAlsa::probeDeviceOpen: error getting minimum channels for device (" << name << "), " << snd_strerror( result ) << ".";\r
-- errorText_ = errorStream_.str();\r
-- return FAILURE;\r
-- }\r
-- deviceChannels = value;\r
-- if ( deviceChannels < channels + firstChannel ) deviceChannels = channels + firstChannel;\r
-- stream_.nDeviceChannels[mode] = deviceChannels;\r
--\r
-- // Set the device channels.\r
-- result = snd_pcm_hw_params_set_channels( phandle, hw_params, deviceChannels );\r
-- if ( result < 0 ) {\r
-- snd_pcm_close( phandle );\r
-- errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting channels for device (" << name << "), " << snd_strerror( result ) << ".";\r
-- errorText_ = errorStream_.str();\r
-- return FAILURE;\r
-- }\r
--\r
-- // Set the buffer (or period) size.\r
-- int dir = 0;\r
-- snd_pcm_uframes_t periodSize = *bufferSize;\r
-- result = snd_pcm_hw_params_set_period_size_near( phandle, hw_params, &periodSize, &dir );\r
-- if ( result < 0 ) {\r
-- snd_pcm_close( phandle );\r
-- errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting period size for device (" << name << "), " << snd_strerror( result ) << ".";\r
-- errorText_ = errorStream_.str();\r
-- return FAILURE;\r
-- }\r
-- *bufferSize = periodSize;\r
--\r
-- // Set the buffer number, which in ALSA is referred to as the "period".\r
-- unsigned int periods = 0;\r
-- if ( options && options->flags & RTAUDIO_MINIMIZE_LATENCY ) periods = 2;\r
-- if ( options && options->numberOfBuffers > 0 ) periods = options->numberOfBuffers;\r
-- if ( periods < 2 ) periods = 4; // a fairly safe default value\r
-- result = snd_pcm_hw_params_set_periods_near( phandle, hw_params, &periods, &dir );\r
-- if ( result < 0 ) {\r
-- snd_pcm_close( phandle );\r
-- errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting periods for device (" << name << "), " << snd_strerror( result ) << ".";\r
-- errorText_ = errorStream_.str();\r
-- return FAILURE;\r
-- }\r
--\r
-- // If attempting to setup a duplex stream, the bufferSize parameter\r
-- // MUST be the same in both directions!\r
-- if ( stream_.mode == OUTPUT && mode == INPUT && *bufferSize != stream_.bufferSize ) {\r
-- snd_pcm_close( phandle );\r
-- errorStream_ << "RtApiAlsa::probeDeviceOpen: system error setting buffer size for duplex stream on device (" << name << ").";\r
-- errorText_ = errorStream_.str();\r
-- return FAILURE;\r
-- }\r
--\r
-- stream_.bufferSize = *bufferSize;\r
--\r
-- // Install the hardware configuration\r
-- result = snd_pcm_hw_params( phandle, hw_params );\r
-- if ( result < 0 ) {\r
-- snd_pcm_close( phandle );\r
-- errorStream_ << "RtApiAlsa::probeDeviceOpen: error installing hardware configuration on device (" << name << "), " << snd_strerror( result ) << ".";\r
-- errorText_ = errorStream_.str();\r
-- return FAILURE;\r
-- }\r
--\r
--#if defined(__RTAUDIO_DEBUG__)\r
-- fprintf(stderr, "\nRtApiAlsa: dump hardware params after installation:\n\n");\r
-- snd_pcm_hw_params_dump( hw_params, out );\r
--#endif\r
--\r
-- // Set the software configuration to fill buffers with zeros and prevent device stopping on xruns.\r
-- snd_pcm_sw_params_t *sw_params = NULL;\r
-- snd_pcm_sw_params_alloca( &sw_params );\r
-- snd_pcm_sw_params_current( phandle, sw_params );\r
-- snd_pcm_sw_params_set_start_threshold( phandle, sw_params, *bufferSize );\r
-- snd_pcm_sw_params_set_stop_threshold( phandle, sw_params, ULONG_MAX );\r
-- snd_pcm_sw_params_set_silence_threshold( phandle, sw_params, 0 );\r
--\r
-- // The following two settings were suggested by Theo Veenker\r
-- //snd_pcm_sw_params_set_avail_min( phandle, sw_params, *bufferSize );\r
-- //snd_pcm_sw_params_set_xfer_align( phandle, sw_params, 1 );\r
--\r
-- // here are two options for a fix\r
-- //snd_pcm_sw_params_set_silence_size( phandle, sw_params, ULONG_MAX );\r
-- snd_pcm_uframes_t val;\r
-- snd_pcm_sw_params_get_boundary( sw_params, &val );\r
-- snd_pcm_sw_params_set_silence_size( phandle, sw_params, val );\r
--\r
-- result = snd_pcm_sw_params( phandle, sw_params );\r
-- if ( result < 0 ) {\r
-- snd_pcm_close( phandle );\r
-- errorStream_ << "RtApiAlsa::probeDeviceOpen: error installing software configuration on device (" << name << "), " << snd_strerror( result ) << ".";\r
-- errorText_ = errorStream_.str();\r
-- return FAILURE;\r
-- }\r
--\r
--#if defined(__RTAUDIO_DEBUG__)\r
-- fprintf(stderr, "\nRtApiAlsa: dump software params after installation:\n\n");\r
-- snd_pcm_sw_params_dump( sw_params, out );\r
--#endif\r
--\r
-- // Set flags for buffer conversion\r
-- stream_.doConvertBuffer[mode] = false;\r
-- if ( stream_.userFormat != stream_.deviceFormat[mode] )\r
-- stream_.doConvertBuffer[mode] = true;\r
-- if ( stream_.nUserChannels[mode] < stream_.nDeviceChannels[mode] )\r
-- stream_.doConvertBuffer[mode] = true;\r
-- if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] &&\r
-- stream_.nUserChannels[mode] > 1 )\r
-- stream_.doConvertBuffer[mode] = true;\r
--\r
-- // Allocate the ApiHandle if necessary and then save.\r
-- AlsaHandle *apiInfo = 0;\r
-- if ( stream_.apiHandle == 0 ) {\r
-- try {\r
-- apiInfo = (AlsaHandle *) new AlsaHandle;\r
-- }\r
-- catch ( std::bad_alloc& ) {\r
-- errorText_ = "RtApiAlsa::probeDeviceOpen: error allocating AlsaHandle memory.";\r
-- goto error;\r
-- }\r
--\r
-- if ( pthread_cond_init( &apiInfo->runnable_cv, NULL ) ) {\r
-- errorText_ = "RtApiAlsa::probeDeviceOpen: error initializing pthread condition variable.";\r
-- goto error;\r
-- }\r
--\r
-- stream_.apiHandle = (void *) apiInfo;\r
-- apiInfo->handles[0] = 0;\r
-- apiInfo->handles[1] = 0;\r
-- }\r
-- else {\r
-- apiInfo = (AlsaHandle *) stream_.apiHandle;\r
-- }\r
-- apiInfo->handles[mode] = phandle;\r
-- phandle = 0;\r
--\r
-- // Allocate necessary internal buffers.\r
-- unsigned long bufferBytes;\r
-- bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat );\r
-- stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 );\r
-- if ( stream_.userBuffer[mode] == NULL ) {\r
-- errorText_ = "RtApiAlsa::probeDeviceOpen: error allocating user buffer memory.";\r
-- goto error;\r
-- }\r
--\r
-- if ( stream_.doConvertBuffer[mode] ) {\r
--\r
-- bool makeBuffer = true;\r
-- bufferBytes = stream_.nDeviceChannels[mode] * formatBytes( stream_.deviceFormat[mode] );\r
-- if ( mode == INPUT ) {\r
-- if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) {\r
-- unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] );\r
-- if ( bufferBytes <= bytesOut ) makeBuffer = false;\r
-- }\r
-- }\r
--\r
-- if ( makeBuffer ) {\r
-- bufferBytes *= *bufferSize;\r
-- if ( stream_.deviceBuffer ) free( stream_.deviceBuffer );\r
-- stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 );\r
-- if ( stream_.deviceBuffer == NULL ) {\r
-- errorText_ = "RtApiAlsa::probeDeviceOpen: error allocating device buffer memory.";\r
-- goto error;\r
-- }\r
-- }\r
-- }\r
--\r
-- stream_.sampleRate = sampleRate;\r
-- stream_.nBuffers = periods;\r
-- stream_.device[mode] = device;\r
-- stream_.state = STREAM_STOPPED;\r
--\r
-- // Setup the buffer conversion information structure.\r
-- if ( stream_.doConvertBuffer[mode] ) setConvertInfo( mode, firstChannel );\r
--\r
-- // Setup thread if necessary.\r
-- if ( stream_.mode == OUTPUT && mode == INPUT ) {\r
-- // We had already set up an output stream.\r
-- stream_.mode = DUPLEX;\r
-- // Link the streams if possible.\r
-- apiInfo->synchronized = false;\r
-- if ( snd_pcm_link( apiInfo->handles[0], apiInfo->handles[1] ) == 0 )\r
-- apiInfo->synchronized = true;\r
-- else {\r
-- errorText_ = "RtApiAlsa::probeDeviceOpen: unable to synchronize input and output devices.";\r
-- error( RtAudioError::WARNING );\r
-- }\r
-- }\r
-- else {\r
-- stream_.mode = mode;\r
--\r
-- // Setup callback thread.\r
-- stream_.callbackInfo.object = (void *) this;\r
--\r
-- // Set the thread attributes for joinable and realtime scheduling\r
-- // priority (optional). The higher priority will only take affect\r
-- // if the program is run as root or suid. Note, under Linux\r
-- // processes with CAP_SYS_NICE privilege, a user can change\r
-- // scheduling policy and priority (thus need not be root). See\r
-- // POSIX "capabilities".\r
-- pthread_attr_t attr;\r
-- pthread_attr_init( &attr );\r
-- pthread_attr_setdetachstate( &attr, PTHREAD_CREATE_JOINABLE );\r
--\r
--#ifdef SCHED_RR // Undefined with some OSes (eg: NetBSD 1.6.x with GNU Pthread)\r
-- if ( options && options->flags & RTAUDIO_SCHEDULE_REALTIME ) {\r
-- // We previously attempted to increase the audio callback priority\r
-- // to SCHED_RR here via the attributes. However, while no errors\r
-- // were reported in doing so, it did not work. So, now this is\r
-- // done in the alsaCallbackHandler function.\r
-- stream_.callbackInfo.doRealtime = true;\r
-- int priority = options->priority;\r
-- int min = sched_get_priority_min( SCHED_RR );\r
-- int max = sched_get_priority_max( SCHED_RR );\r
-- if ( priority < min ) priority = min;\r
-- else if ( priority > max ) priority = max;\r
-- stream_.callbackInfo.priority = priority;\r
-- }\r
--#endif\r
--\r
-- stream_.callbackInfo.isRunning = true;\r
-- result = pthread_create( &stream_.callbackInfo.thread, &attr, alsaCallbackHandler, &stream_.callbackInfo );\r
-- pthread_attr_destroy( &attr );\r
-- if ( result ) {\r
-- stream_.callbackInfo.isRunning = false;\r
-- errorText_ = "RtApiAlsa::error creating callback thread!";\r
-- goto error;\r
-- }\r
-- }\r
--\r
-- return SUCCESS;\r
--\r
-- error:\r
-- if ( apiInfo ) {\r
-- pthread_cond_destroy( &apiInfo->runnable_cv );\r
-- if ( apiInfo->handles[0] ) snd_pcm_close( apiInfo->handles[0] );\r
-- if ( apiInfo->handles[1] ) snd_pcm_close( apiInfo->handles[1] );\r
-- delete apiInfo;\r
-- stream_.apiHandle = 0;\r
-- }\r
--\r
-- if ( phandle) snd_pcm_close( phandle );\r
--\r
-- for ( int i=0; i<2; i++ ) {\r
-- if ( stream_.userBuffer[i] ) {\r
-- free( stream_.userBuffer[i] );\r
-- stream_.userBuffer[i] = 0;\r
-- }\r
-- }\r
--\r
-- if ( stream_.deviceBuffer ) {\r
-- free( stream_.deviceBuffer );\r
-- stream_.deviceBuffer = 0;\r
-- }\r
--\r
-- stream_.state = STREAM_CLOSED;\r
-- return FAILURE;\r
--}\r
--\r
--void RtApiAlsa :: closeStream()\r
--{\r
-- if ( stream_.state == STREAM_CLOSED ) {\r
-- errorText_ = "RtApiAlsa::closeStream(): no open stream to close!";\r
-- error( RtAudioError::WARNING );\r
-- return;\r
-- }\r
--\r
-- AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle;\r
-- stream_.callbackInfo.isRunning = false;\r
-- MUTEX_LOCK( &stream_.mutex );\r
-- if ( stream_.state == STREAM_STOPPED ) {\r
-- apiInfo->runnable = true;\r
-- pthread_cond_signal( &apiInfo->runnable_cv );\r
-- }\r
-- MUTEX_UNLOCK( &stream_.mutex );\r
-- pthread_join( stream_.callbackInfo.thread, NULL );\r
--\r
-- if ( stream_.state == STREAM_RUNNING ) {\r
-- stream_.state = STREAM_STOPPED;\r
-- if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX )\r
-- snd_pcm_drop( apiInfo->handles[0] );\r
-- if ( stream_.mode == INPUT || stream_.mode == DUPLEX )\r
-- snd_pcm_drop( apiInfo->handles[1] );\r
-- }\r
--\r
-- if ( apiInfo ) {\r
-- pthread_cond_destroy( &apiInfo->runnable_cv );\r
-- if ( apiInfo->handles[0] ) snd_pcm_close( apiInfo->handles[0] );\r
-- if ( apiInfo->handles[1] ) snd_pcm_close( apiInfo->handles[1] );\r
-- delete apiInfo;\r
-- stream_.apiHandle = 0;\r
-- }\r
--\r
-- for ( int i=0; i<2; i++ ) {\r
-- if ( stream_.userBuffer[i] ) {\r
-- free( stream_.userBuffer[i] );\r
-- stream_.userBuffer[i] = 0;\r
-- }\r
-- }\r
--\r
-- if ( stream_.deviceBuffer ) {\r
-- free( stream_.deviceBuffer );\r
-- stream_.deviceBuffer = 0;\r
-- }\r
--\r
-- stream_.mode = UNINITIALIZED;\r
-- stream_.state = STREAM_CLOSED;\r
--}\r
--\r
--void RtApiAlsa :: startStream()\r
--{\r
-- // This method calls snd_pcm_prepare if the device isn't already in that state.\r
--\r
-- verifyStream();\r
-- if ( stream_.state == STREAM_RUNNING ) {\r
-- errorText_ = "RtApiAlsa::startStream(): the stream is already running!";\r
-- error( RtAudioError::WARNING );\r
-- return;\r
-- }\r
--\r
-- MUTEX_LOCK( &stream_.mutex );\r
--\r
-- int result = 0;\r
-- snd_pcm_state_t state;\r
-- AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle;\r
-- snd_pcm_t **handle = (snd_pcm_t **) apiInfo->handles;\r
-- if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {\r
-- state = snd_pcm_state( handle[0] );\r
-- if ( state != SND_PCM_STATE_PREPARED ) {\r
-- result = snd_pcm_prepare( handle[0] );\r
-- if ( result < 0 ) {\r
-- errorStream_ << "RtApiAlsa::startStream: error preparing output pcm device, " << snd_strerror( result ) << ".";\r
-- errorText_ = errorStream_.str();\r
-- goto unlock;\r
-- }\r
-- }\r
-- }\r
--\r
-- if ( ( stream_.mode == INPUT || stream_.mode == DUPLEX ) && !apiInfo->synchronized ) {\r
-- result = snd_pcm_drop(handle[1]); // fix to remove stale data received since device has been open\r
-- state = snd_pcm_state( handle[1] );\r
-- if ( state != SND_PCM_STATE_PREPARED ) {\r
-- result = snd_pcm_prepare( handle[1] );\r
-- if ( result < 0 ) {\r
-- errorStream_ << "RtApiAlsa::startStream: error preparing input pcm device, " << snd_strerror( result ) << ".";\r
-- errorText_ = errorStream_.str();\r
-- goto unlock;\r
-- }\r
-- }\r
-- }\r
--\r
-- stream_.state = STREAM_RUNNING;\r
--\r
-- unlock:\r
-- apiInfo->runnable = true;\r
-- pthread_cond_signal( &apiInfo->runnable_cv );\r
-- MUTEX_UNLOCK( &stream_.mutex );\r
--\r
-- if ( result >= 0 ) return;\r
-- error( RtAudioError::SYSTEM_ERROR );\r
--}\r
--\r
--void RtApiAlsa :: stopStream()\r
--{\r
-- verifyStream();\r
-- if ( stream_.state == STREAM_STOPPED ) {\r
-- errorText_ = "RtApiAlsa::stopStream(): the stream is already stopped!";\r
-- error( RtAudioError::WARNING );\r
-- return;\r
-- }\r
--\r
-- stream_.state = STREAM_STOPPED;\r
-- MUTEX_LOCK( &stream_.mutex );\r
--\r
-- int result = 0;\r
-- AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle;\r
-- snd_pcm_t **handle = (snd_pcm_t **) apiInfo->handles;\r
-- if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {\r
-- if ( apiInfo->synchronized )\r
-- result = snd_pcm_drop( handle[0] );\r
-- else\r
-- result = snd_pcm_drain( handle[0] );\r
-- if ( result < 0 ) {\r
-- errorStream_ << "RtApiAlsa::stopStream: error draining output pcm device, " << snd_strerror( result ) << ".";\r
-- errorText_ = errorStream_.str();\r
-- goto unlock;\r
-- }\r
-- }\r
--\r
-- if ( ( stream_.mode == INPUT || stream_.mode == DUPLEX ) && !apiInfo->synchronized ) {\r
-- result = snd_pcm_drop( handle[1] );\r
-- if ( result < 0 ) {\r
-- errorStream_ << "RtApiAlsa::stopStream: error stopping input pcm device, " << snd_strerror( result ) << ".";\r
-- errorText_ = errorStream_.str();\r
-- goto unlock;\r
-- }\r
-- }\r
--\r
-- unlock:\r
-- apiInfo->runnable = false; // fixes high CPU usage when stopped\r
-- MUTEX_UNLOCK( &stream_.mutex );\r
--\r
-- if ( result >= 0 ) return;\r
-- error( RtAudioError::SYSTEM_ERROR );\r
--}\r
--\r
--void RtApiAlsa :: abortStream()\r
--{\r
-- verifyStream();\r
-- if ( stream_.state == STREAM_STOPPED ) {\r
-- errorText_ = "RtApiAlsa::abortStream(): the stream is already stopped!";\r
-- error( RtAudioError::WARNING );\r
-- return;\r
-- }\r
--\r
-- stream_.state = STREAM_STOPPED;\r
-- MUTEX_LOCK( &stream_.mutex );\r
--\r
-- int result = 0;\r
-- AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle;\r
-- snd_pcm_t **handle = (snd_pcm_t **) apiInfo->handles;\r
-- if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {\r
-- result = snd_pcm_drop( handle[0] );\r
-- if ( result < 0 ) {\r
-- errorStream_ << "RtApiAlsa::abortStream: error aborting output pcm device, " << snd_strerror( result ) << ".";\r
-- errorText_ = errorStream_.str();\r
-- goto unlock;\r
-- }\r
-- }\r
--\r
-- if ( ( stream_.mode == INPUT || stream_.mode == DUPLEX ) && !apiInfo->synchronized ) {\r
-- result = snd_pcm_drop( handle[1] );\r
-- if ( result < 0 ) {\r
-- errorStream_ << "RtApiAlsa::abortStream: error aborting input pcm device, " << snd_strerror( result ) << ".";\r
-- errorText_ = errorStream_.str();\r
-- goto unlock;\r
-- }\r
-- }\r
--\r
-- unlock:\r
-- apiInfo->runnable = false; // fixes high CPU usage when stopped\r
-- MUTEX_UNLOCK( &stream_.mutex );\r
--\r
-- if ( result >= 0 ) return;\r
-- error( RtAudioError::SYSTEM_ERROR );\r
--}\r
--\r
--void RtApiAlsa :: callbackEvent()\r
--{\r
-- AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle;\r
-- if ( stream_.state == STREAM_STOPPED ) {\r
-- MUTEX_LOCK( &stream_.mutex );\r
-- while ( !apiInfo->runnable )\r
-- pthread_cond_wait( &apiInfo->runnable_cv, &stream_.mutex );\r
--\r
-- if ( stream_.state != STREAM_RUNNING ) {\r
-- MUTEX_UNLOCK( &stream_.mutex );\r
-- return;\r
-- }\r
-- MUTEX_UNLOCK( &stream_.mutex );\r
-- }\r
--\r
-- if ( stream_.state == STREAM_CLOSED ) {\r
-- errorText_ = "RtApiAlsa::callbackEvent(): the stream is closed ... this shouldn't happen!";\r
-- error( RtAudioError::WARNING );\r
-- return;\r
-- }\r
--\r
-- int doStopStream = 0;\r
-- RtAudioCallback callback = (RtAudioCallback) stream_.callbackInfo.callback;\r
-- double streamTime = getStreamTime();\r
-- RtAudioStreamStatus status = 0;\r
-- if ( stream_.mode != INPUT && apiInfo->xrun[0] == true ) {\r
-- status |= RTAUDIO_OUTPUT_UNDERFLOW;\r
-- apiInfo->xrun[0] = false;\r
-- }\r
-- if ( stream_.mode != OUTPUT && apiInfo->xrun[1] == true ) {\r
-- status |= RTAUDIO_INPUT_OVERFLOW;\r
-- apiInfo->xrun[1] = false;\r
-- }\r
-- doStopStream = callback( stream_.userBuffer[0], stream_.userBuffer[1],\r
-- stream_.bufferSize, streamTime, status, stream_.callbackInfo.userData );\r
--\r
-- if ( doStopStream == 2 ) {\r
-- abortStream();\r
-- return;\r
-- }\r
--\r
-- MUTEX_LOCK( &stream_.mutex );\r
--\r
-- // The state might change while waiting on a mutex.\r
-- if ( stream_.state == STREAM_STOPPED ) goto unlock;\r
--\r
-- int result;\r
-- char *buffer;\r
-- int channels;\r
-- snd_pcm_t **handle;\r
-- snd_pcm_sframes_t frames;\r
-- RtAudioFormat format;\r
-- handle = (snd_pcm_t **) apiInfo->handles;\r
--\r
-- if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {\r
--\r
-- // Setup parameters.\r
-- if ( stream_.doConvertBuffer[1] ) {\r
-- buffer = stream_.deviceBuffer;\r
-- channels = stream_.nDeviceChannels[1];\r
-- format = stream_.deviceFormat[1];\r
-- }\r
-- else {\r
-- buffer = stream_.userBuffer[1];\r
-- channels = stream_.nUserChannels[1];\r
-- format = stream_.userFormat;\r
-- }\r
--\r
-- // Read samples from device in interleaved/non-interleaved format.\r
-- if ( stream_.deviceInterleaved[1] )\r
-- result = snd_pcm_readi( handle[1], buffer, stream_.bufferSize );\r
-- else {\r
-- void *bufs[channels];\r
-- size_t offset = stream_.bufferSize * formatBytes( format );\r
-- for ( int i=0; i<channels; i++ )\r
-- bufs[i] = (void *) (buffer + (i * offset));\r
-- result = snd_pcm_readn( handle[1], bufs, stream_.bufferSize );\r
-- }\r
--\r
-- if ( result < (int) stream_.bufferSize ) {\r
-- // Either an error or overrun occured.\r
-- if ( result == -EPIPE ) {\r
-- snd_pcm_state_t state = snd_pcm_state( handle[1] );\r
-- if ( state == SND_PCM_STATE_XRUN ) {\r
-- apiInfo->xrun[1] = true;\r
-- result = snd_pcm_prepare( handle[1] );\r
-- if ( result < 0 ) {\r
-- errorStream_ << "RtApiAlsa::callbackEvent: error preparing device after overrun, " << snd_strerror( result ) << ".";\r
-- errorText_ = errorStream_.str();\r
-- }\r
-- }\r
-- else {\r
-- errorStream_ << "RtApiAlsa::callbackEvent: error, current state is " << snd_pcm_state_name( state ) << ", " << snd_strerror( result ) << ".";\r
-- errorText_ = errorStream_.str();\r
-- }\r
-- }\r
-- else {\r
-- errorStream_ << "RtApiAlsa::callbackEvent: audio read error, " << snd_strerror( result ) << ".";\r
-- errorText_ = errorStream_.str();\r
-- }\r
-- error( RtAudioError::WARNING );\r
-- goto tryOutput;\r
-- }\r
--\r
-- // Do byte swapping if necessary.\r
-- if ( stream_.doByteSwap[1] )\r
-- byteSwapBuffer( buffer, stream_.bufferSize * channels, format );\r
--\r
-- // Do buffer conversion if necessary.\r
-- if ( stream_.doConvertBuffer[1] )\r
-- convertBuffer( stream_.userBuffer[1], stream_.deviceBuffer, stream_.convertInfo[1] );\r
--\r
-- // Check stream latency\r
-- result = snd_pcm_delay( handle[1], &frames );\r
-- if ( result == 0 && frames > 0 ) stream_.latency[1] = frames;\r
-- }\r
--\r
-- tryOutput:\r
--\r
-- if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {\r
--\r
-- // Setup parameters and do buffer conversion if necessary.\r
-- if ( stream_.doConvertBuffer[0] ) {\r
-- buffer = stream_.deviceBuffer;\r
-- convertBuffer( buffer, stream_.userBuffer[0], stream_.convertInfo[0] );\r
-- channels = stream_.nDeviceChannels[0];\r
-- format = stream_.deviceFormat[0];\r
-- }\r
-- else {\r
-- buffer = stream_.userBuffer[0];\r
-- channels = stream_.nUserChannels[0];\r
-- format = stream_.userFormat;\r
-- }\r
--\r
-- // Do byte swapping if necessary.\r
-- if ( stream_.doByteSwap[0] )\r
-- byteSwapBuffer(buffer, stream_.bufferSize * channels, format);\r
--\r
-- // Write samples to device in interleaved/non-interleaved format.\r
-- if ( stream_.deviceInterleaved[0] )\r
-- result = snd_pcm_writei( handle[0], buffer, stream_.bufferSize );\r
-- else {\r
-- void *bufs[channels];\r
-- size_t offset = stream_.bufferSize * formatBytes( format );\r
-- for ( int i=0; i<channels; i++ )\r
-- bufs[i] = (void *) (buffer + (i * offset));\r
-- result = snd_pcm_writen( handle[0], bufs, stream_.bufferSize );\r
-- }\r
--\r
-- if ( result < (int) stream_.bufferSize ) {\r
-- // Either an error or underrun occured.\r
-- if ( result == -EPIPE ) {\r
-- snd_pcm_state_t state = snd_pcm_state( handle[0] );\r
-- if ( state == SND_PCM_STATE_XRUN ) {\r
-- apiInfo->xrun[0] = true;\r
-- result = snd_pcm_prepare( handle[0] );\r
-- if ( result < 0 ) {\r
-- errorStream_ << "RtApiAlsa::callbackEvent: error preparing device after underrun, " << snd_strerror( result ) << ".";\r
-- errorText_ = errorStream_.str();\r
-- }\r
-- else\r
-- errorText_ = "RtApiAlsa::callbackEvent: audio write error, underrun.";\r
-- }\r
-- else {\r
-- errorStream_ << "RtApiAlsa::callbackEvent: error, current state is " << snd_pcm_state_name( state ) << ", " << snd_strerror( result ) << ".";\r
-- errorText_ = errorStream_.str();\r
-- }\r
-- }\r
-- else {\r
-- errorStream_ << "RtApiAlsa::callbackEvent: audio write error, " << snd_strerror( result ) << ".";\r
-- errorText_ = errorStream_.str();\r
-- }\r
-- error( RtAudioError::WARNING );\r
-- goto unlock;\r
-- }\r
--\r
-- // Check stream latency\r
-- result = snd_pcm_delay( handle[0], &frames );\r
-- if ( result == 0 && frames > 0 ) stream_.latency[0] = frames;\r
-- }\r
--\r
-- unlock:\r
-- MUTEX_UNLOCK( &stream_.mutex );\r
--\r
-- RtApi::tickStreamTime();\r
-- if ( doStopStream == 1 ) this->stopStream();\r
--}\r
--\r
--static void *alsaCallbackHandler( void *ptr )\r
--{\r
-- CallbackInfo *info = (CallbackInfo *) ptr;\r
-- RtApiAlsa *object = (RtApiAlsa *) info->object;\r
-- bool *isRunning = &info->isRunning;\r
--\r
--#ifdef SCHED_RR // Undefined with some OSes (eg: NetBSD 1.6.x with GNU Pthread)\r
-- if ( info->doRealtime ) {\r
-- pthread_t tID = pthread_self(); // ID of this thread\r
-- sched_param prio = { info->priority }; // scheduling priority of thread\r
-- pthread_setschedparam( tID, SCHED_RR, &prio );\r
-- }\r
--#endif\r
--\r
-- while ( *isRunning == true ) {\r
-- pthread_testcancel();\r
-- object->callbackEvent();\r
-- }\r
--\r
-- pthread_exit( NULL );\r
--}\r
--\r
--//******************** End of __LINUX_ALSA__ *********************//\r
--#endif\r
--\r
--#if defined(__LINUX_PULSE__)\r
--\r
--// Code written by Peter Meerwald, pmeerw@pmeerw.net\r
--// and Tristan Matthews.\r
--\r
--#include <pulse/error.h>\r
--#include <pulse/simple.h>\r
--#include <cstdio>\r
--\r
--static const unsigned int SUPPORTED_SAMPLERATES[] = { 8000, 16000, 22050, 32000,\r
-- 44100, 48000, 96000, 0};\r
--\r
--struct rtaudio_pa_format_mapping_t {\r
-- RtAudioFormat rtaudio_format;\r
-- pa_sample_format_t pa_format;\r
--};\r
--\r
--static const rtaudio_pa_format_mapping_t supported_sampleformats[] = {\r
-- {RTAUDIO_SINT16, PA_SAMPLE_S16LE},\r
-- {RTAUDIO_SINT32, PA_SAMPLE_S32LE},\r
-- {RTAUDIO_FLOAT32, PA_SAMPLE_FLOAT32LE},\r
-- {0, PA_SAMPLE_INVALID}};\r
--\r
--struct PulseAudioHandle {\r
-- pa_simple *s_play;\r
-- pa_simple *s_rec;\r
-- pthread_t thread;\r
-- pthread_cond_t runnable_cv;\r
-- bool runnable;\r
-- PulseAudioHandle() : s_play(0), s_rec(0), runnable(false) { }\r
--};\r
--\r
--RtApiPulse::~RtApiPulse()\r
--{\r
-- if ( stream_.state != STREAM_CLOSED )\r
-- closeStream();\r
--}\r
--\r
--unsigned int RtApiPulse::getDeviceCount( void )\r
--{\r
-- return 1;\r
--}\r
--\r
--RtAudio::DeviceInfo RtApiPulse::getDeviceInfo( unsigned int /*device*/ )\r
--{\r
-- RtAudio::DeviceInfo info;\r
-- info.probed = true;\r
-- info.name = "PulseAudio";\r
-- info.outputChannels = 2;\r
-- info.inputChannels = 2;\r
-- info.duplexChannels = 2;\r
-- info.isDefaultOutput = true;\r
-- info.isDefaultInput = true;\r
--\r
-- for ( const unsigned int *sr = SUPPORTED_SAMPLERATES; *sr; ++sr )\r
-- info.sampleRates.push_back( *sr );\r
--\r
-- info.preferredSampleRate = 48000;\r
-- info.nativeFormats = RTAUDIO_SINT16 | RTAUDIO_SINT32 | RTAUDIO_FLOAT32;\r
--\r
-- return info;\r
--}\r
--\r
--static void *pulseaudio_callback( void * user )\r
--{\r
-- CallbackInfo *cbi = static_cast<CallbackInfo *>( user );\r
-- RtApiPulse *context = static_cast<RtApiPulse *>( cbi->object );\r
-- volatile bool *isRunning = &cbi->isRunning;\r
--\r
-- while ( *isRunning ) {\r
-- pthread_testcancel();\r
-- context->callbackEvent();\r
-- }\r
--\r
-- pthread_exit( NULL );\r
--}\r
--\r
--void RtApiPulse::closeStream( void )\r
--{\r
-- PulseAudioHandle *pah = static_cast<PulseAudioHandle *>( stream_.apiHandle );\r
--\r
-- stream_.callbackInfo.isRunning = false;\r
-- if ( pah ) {\r
-- MUTEX_LOCK( &stream_.mutex );\r
-- if ( stream_.state == STREAM_STOPPED ) {\r
-- pah->runnable = true;\r
-- pthread_cond_signal( &pah->runnable_cv );\r
-- }\r
-- MUTEX_UNLOCK( &stream_.mutex );\r
--\r
-- pthread_join( pah->thread, 0 );\r
-- if ( pah->s_play ) {\r
-- pa_simple_flush( pah->s_play, NULL );\r
-- pa_simple_free( pah->s_play );\r
-- }\r
-- if ( pah->s_rec )\r
-- pa_simple_free( pah->s_rec );\r
--\r
-- pthread_cond_destroy( &pah->runnable_cv );\r
-- delete pah;\r
-- stream_.apiHandle = 0;\r
-- }\r
--\r
-- if ( stream_.userBuffer[0] ) {\r
-- free( stream_.userBuffer[0] );\r
-- stream_.userBuffer[0] = 0;\r
-- }\r
-- if ( stream_.userBuffer[1] ) {\r
-- free( stream_.userBuffer[1] );\r
-- stream_.userBuffer[1] = 0;\r
-- }\r
--\r
-- stream_.state = STREAM_CLOSED;\r
-- stream_.mode = UNINITIALIZED;\r
--}\r
--\r
--void RtApiPulse::callbackEvent( void )\r
--{\r
-- PulseAudioHandle *pah = static_cast<PulseAudioHandle *>( stream_.apiHandle );\r
--\r
-- if ( stream_.state == STREAM_STOPPED ) {\r
-- MUTEX_LOCK( &stream_.mutex );\r
-- while ( !pah->runnable )\r
-- pthread_cond_wait( &pah->runnable_cv, &stream_.mutex );\r
--\r
-- if ( stream_.state != STREAM_RUNNING ) {\r
-- MUTEX_UNLOCK( &stream_.mutex );\r
-- return;\r
-- }\r
-- MUTEX_UNLOCK( &stream_.mutex );\r
-- }\r
--\r
-- if ( stream_.state == STREAM_CLOSED ) {\r
-- errorText_ = "RtApiPulse::callbackEvent(): the stream is closed ... "\r
-- "this shouldn't happen!";\r
-- error( RtAudioError::WARNING );\r
-- return;\r
-- }\r
--\r
-- RtAudioCallback callback = (RtAudioCallback) stream_.callbackInfo.callback;\r
-- double streamTime = getStreamTime();\r
-- RtAudioStreamStatus status = 0;\r
-- int doStopStream = callback( stream_.userBuffer[OUTPUT], stream_.userBuffer[INPUT],\r
-- stream_.bufferSize, streamTime, status,\r
-- stream_.callbackInfo.userData );\r
--\r
-- if ( doStopStream == 2 ) {\r
-- abortStream();\r
-- return;\r
-- }\r
--\r
-- MUTEX_LOCK( &stream_.mutex );\r
-- void *pulse_in = stream_.doConvertBuffer[INPUT] ? stream_.deviceBuffer : stream_.userBuffer[INPUT];\r
-- void *pulse_out = stream_.doConvertBuffer[OUTPUT] ? stream_.deviceBuffer : stream_.userBuffer[OUTPUT];\r
--\r
-- if ( stream_.state != STREAM_RUNNING )\r
-- goto unlock;\r
--\r
-- int pa_error;\r
-- size_t bytes;\r
-- if (stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {\r
-- if ( stream_.doConvertBuffer[OUTPUT] ) {\r
-- convertBuffer( stream_.deviceBuffer,\r
-- stream_.userBuffer[OUTPUT],\r
-- stream_.convertInfo[OUTPUT] );\r
-- bytes = stream_.nDeviceChannels[OUTPUT] * stream_.bufferSize *\r
-- formatBytes( stream_.deviceFormat[OUTPUT] );\r
-- } else\r
-- bytes = stream_.nUserChannels[OUTPUT] * stream_.bufferSize *\r
-- formatBytes( stream_.userFormat );\r
--\r
-- if ( pa_simple_write( pah->s_play, pulse_out, bytes, &pa_error ) < 0 ) {\r
-- errorStream_ << "RtApiPulse::callbackEvent: audio write error, " <<\r
-- pa_strerror( pa_error ) << ".";\r
-- errorText_ = errorStream_.str();\r
-- error( RtAudioError::WARNING );\r
-- }\r
-- }\r
--\r
-- if ( stream_.mode == INPUT || stream_.mode == DUPLEX) {\r
-- if ( stream_.doConvertBuffer[INPUT] )\r
-- bytes = stream_.nDeviceChannels[INPUT] * stream_.bufferSize *\r
-- formatBytes( stream_.deviceFormat[INPUT] );\r
-- else\r
-- bytes = stream_.nUserChannels[INPUT] * stream_.bufferSize *\r
-- formatBytes( stream_.userFormat );\r
--\r
-- if ( pa_simple_read( pah->s_rec, pulse_in, bytes, &pa_error ) < 0 ) {\r
-- errorStream_ << "RtApiPulse::callbackEvent: audio read error, " <<\r
-- pa_strerror( pa_error ) << ".";\r
-- errorText_ = errorStream_.str();\r
-- error( RtAudioError::WARNING );\r
-- }\r
-- if ( stream_.doConvertBuffer[INPUT] ) {\r
-- convertBuffer( stream_.userBuffer[INPUT],\r
-- stream_.deviceBuffer,\r
-- stream_.convertInfo[INPUT] );\r
-- }\r
-- }\r
--\r
-- unlock:\r
-- MUTEX_UNLOCK( &stream_.mutex );\r
-- RtApi::tickStreamTime();\r
--\r
-- if ( doStopStream == 1 )\r
-- stopStream();\r
--}\r
--\r
--void RtApiPulse::startStream( void )\r
--{\r
-- PulseAudioHandle *pah = static_cast<PulseAudioHandle *>( stream_.apiHandle );\r
--\r
-- if ( stream_.state == STREAM_CLOSED ) {\r
-- errorText_ = "RtApiPulse::startStream(): the stream is not open!";\r
-- error( RtAudioError::INVALID_USE );\r
-- return;\r
-- }\r
-- if ( stream_.state == STREAM_RUNNING ) {\r
-- errorText_ = "RtApiPulse::startStream(): the stream is already running!";\r
-- error( RtAudioError::WARNING );\r
-- return;\r
-- }\r
--\r
-- MUTEX_LOCK( &stream_.mutex );\r
--\r
-- stream_.state = STREAM_RUNNING;\r
--\r
-- pah->runnable = true;\r
-- pthread_cond_signal( &pah->runnable_cv );\r
-- MUTEX_UNLOCK( &stream_.mutex );\r
--}\r
--\r
--void RtApiPulse::stopStream( void )\r
--{\r
-- PulseAudioHandle *pah = static_cast<PulseAudioHandle *>( stream_.apiHandle );\r
--\r
-- if ( stream_.state == STREAM_CLOSED ) {\r
-- errorText_ = "RtApiPulse::stopStream(): the stream is not open!";\r
-- error( RtAudioError::INVALID_USE );\r
-- return;\r
-- }\r
-- if ( stream_.state == STREAM_STOPPED ) {\r
-- errorText_ = "RtApiPulse::stopStream(): the stream is already stopped!";\r
-- error( RtAudioError::WARNING );\r
-- return;\r
-- }\r
--\r
-- stream_.state = STREAM_STOPPED;\r
-- MUTEX_LOCK( &stream_.mutex );\r
--\r
-- if ( pah && pah->s_play ) {\r
-- int pa_error;\r
-- if ( pa_simple_drain( pah->s_play, &pa_error ) < 0 ) {\r
-- errorStream_ << "RtApiPulse::stopStream: error draining output device, " <<\r
-- pa_strerror( pa_error ) << ".";\r
-- errorText_ = errorStream_.str();\r
-- MUTEX_UNLOCK( &stream_.mutex );\r
-- error( RtAudioError::SYSTEM_ERROR );\r
-- return;\r
-- }\r
-- }\r
--\r
-- stream_.state = STREAM_STOPPED;\r
-- MUTEX_UNLOCK( &stream_.mutex );\r
--}\r
--\r
--void RtApiPulse::abortStream( void )\r
--{\r
-- PulseAudioHandle *pah = static_cast<PulseAudioHandle*>( stream_.apiHandle );\r
--\r
-- if ( stream_.state == STREAM_CLOSED ) {\r
-- errorText_ = "RtApiPulse::abortStream(): the stream is not open!";\r
-- error( RtAudioError::INVALID_USE );\r
-- return;\r
-- }\r
-- if ( stream_.state == STREAM_STOPPED ) {\r
-- errorText_ = "RtApiPulse::abortStream(): the stream is already stopped!";\r
-- error( RtAudioError::WARNING );\r
-- return;\r
-- }\r
--\r
-- stream_.state = STREAM_STOPPED;\r
-- MUTEX_LOCK( &stream_.mutex );\r
--\r
-- if ( pah && pah->s_play ) {\r
-- int pa_error;\r
-- if ( pa_simple_flush( pah->s_play, &pa_error ) < 0 ) {\r
-- errorStream_ << "RtApiPulse::abortStream: error flushing output device, " <<\r
-- pa_strerror( pa_error ) << ".";\r
-- errorText_ = errorStream_.str();\r
-- MUTEX_UNLOCK( &stream_.mutex );\r
-- error( RtAudioError::SYSTEM_ERROR );\r
-- return;\r
-- }\r
-- }\r
--\r
-- stream_.state = STREAM_STOPPED;\r
-- MUTEX_UNLOCK( &stream_.mutex );\r
--}\r
--\r
--bool RtApiPulse::probeDeviceOpen( unsigned int device, StreamMode mode,\r
-- unsigned int channels, unsigned int firstChannel,\r
-- unsigned int sampleRate, RtAudioFormat format,\r
-- unsigned int *bufferSize, RtAudio::StreamOptions *options )\r
--{\r
-- PulseAudioHandle *pah = 0;\r
-- unsigned long bufferBytes = 0;\r
-- pa_sample_spec ss;\r
--\r
-- if ( device != 0 ) return false;\r
-- if ( mode != INPUT && mode != OUTPUT ) return false;\r
-- if ( channels != 1 && channels != 2 ) {\r
-- errorText_ = "RtApiPulse::probeDeviceOpen: unsupported number of channels.";\r
-- return false;\r
-- }\r
-- ss.channels = channels;\r
--\r
-- if ( firstChannel != 0 ) return false;\r
--\r
-- bool sr_found = false;\r
-- for ( const unsigned int *sr = SUPPORTED_SAMPLERATES; *sr; ++sr ) {\r
-- if ( sampleRate == *sr ) {\r
-- sr_found = true;\r
-- stream_.sampleRate = sampleRate;\r
-- ss.rate = sampleRate;\r
-- break;\r
-- }\r
-- }\r
-- if ( !sr_found ) {\r
-- errorText_ = "RtApiPulse::probeDeviceOpen: unsupported sample rate.";\r
-- return false;\r
-- }\r
--\r
-- bool sf_found = 0;\r
-- for ( const rtaudio_pa_format_mapping_t *sf = supported_sampleformats;\r
-- sf->rtaudio_format && sf->pa_format != PA_SAMPLE_INVALID; ++sf ) {\r
-- if ( format == sf->rtaudio_format ) {\r
-- sf_found = true;\r
-- stream_.userFormat = sf->rtaudio_format;\r
-- stream_.deviceFormat[mode] = stream_.userFormat;\r
-- ss.format = sf->pa_format;\r
-- break;\r
-- }\r
-- }\r
-- if ( !sf_found ) { // Use internal data format conversion.\r
-- stream_.userFormat = format;\r
-- stream_.deviceFormat[mode] = RTAUDIO_FLOAT32;\r
-- ss.format = PA_SAMPLE_FLOAT32LE;\r
-- }\r
--\r
-- // Set other stream parameters.\r
-- if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) stream_.userInterleaved = false;\r
-- else stream_.userInterleaved = true;\r
-- stream_.deviceInterleaved[mode] = true;\r
-- stream_.nBuffers = 1;\r
-- stream_.doByteSwap[mode] = false;\r
-- stream_.nUserChannels[mode] = channels;\r
-- stream_.nDeviceChannels[mode] = channels + firstChannel;\r
-- stream_.channelOffset[mode] = 0;\r
-- std::string streamName = "RtAudio";\r
--\r
-- // Set flags for buffer conversion.\r
-- stream_.doConvertBuffer[mode] = false;\r
-- if ( stream_.userFormat != stream_.deviceFormat[mode] )\r
-- stream_.doConvertBuffer[mode] = true;\r
-- if ( stream_.nUserChannels[mode] < stream_.nDeviceChannels[mode] )\r
-- stream_.doConvertBuffer[mode] = true;\r
--\r
-- // Allocate necessary internal buffers.\r
-- bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat );\r
-- stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 );\r
-- if ( stream_.userBuffer[mode] == NULL ) {\r
-- errorText_ = "RtApiPulse::probeDeviceOpen: error allocating user buffer memory.";\r
-- goto error;\r
-- }\r
-- stream_.bufferSize = *bufferSize;\r
--\r
-- if ( stream_.doConvertBuffer[mode] ) {\r
--\r
-- bool makeBuffer = true;\r
-- bufferBytes = stream_.nDeviceChannels[mode] * formatBytes( stream_.deviceFormat[mode] );\r
-- if ( mode == INPUT ) {\r
-- if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) {\r
-- unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] );\r
-- if ( bufferBytes <= bytesOut ) makeBuffer = false;\r
-- }\r
-- }\r
--\r
-- if ( makeBuffer ) {\r
-- bufferBytes *= *bufferSize;\r
-- if ( stream_.deviceBuffer ) free( stream_.deviceBuffer );\r
-- stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 );\r
-- if ( stream_.deviceBuffer == NULL ) {\r
-- errorText_ = "RtApiPulse::probeDeviceOpen: error allocating device buffer memory.";\r
-- goto error;\r
-- }\r
-- }\r
-- }\r
--\r
-- stream_.device[mode] = device;\r
--\r
-- // Setup the buffer conversion information structure.\r
-- if ( stream_.doConvertBuffer[mode] ) setConvertInfo( mode, firstChannel );\r
--\r
-- if ( !stream_.apiHandle ) {\r
-- PulseAudioHandle *pah = new PulseAudioHandle;\r
-- if ( !pah ) {\r
-- errorText_ = "RtApiPulse::probeDeviceOpen: error allocating memory for handle.";\r
-- goto error;\r
-- }\r
--\r
-- stream_.apiHandle = pah;\r
-- if ( pthread_cond_init( &pah->runnable_cv, NULL ) != 0 ) {\r
-- errorText_ = "RtApiPulse::probeDeviceOpen: error creating condition variable.";\r
-- goto error;\r
-- }\r
-- }\r
-- pah = static_cast<PulseAudioHandle *>( stream_.apiHandle );\r
--\r
-- int error;\r
-- if ( options && !options->streamName.empty() ) streamName = options->streamName;\r
-- switch ( mode ) {\r
-- case INPUT:\r
-- pa_buffer_attr buffer_attr;\r
-- buffer_attr.fragsize = bufferBytes;\r
-- buffer_attr.maxlength = -1;\r
--\r
-- pah->s_rec = pa_simple_new( NULL, streamName.c_str(), PA_STREAM_RECORD, NULL, "Record", &ss, NULL, &buffer_attr, &error );\r
-- if ( !pah->s_rec ) {\r
-- errorText_ = "RtApiPulse::probeDeviceOpen: error connecting input to PulseAudio server.";\r
-- goto error;\r
-- }\r
-- break;\r
-- case OUTPUT:\r
-- pah->s_play = pa_simple_new( NULL, streamName.c_str(), PA_STREAM_PLAYBACK, NULL, "Playback", &ss, NULL, NULL, &error );\r
-- if ( !pah->s_play ) {\r
-- errorText_ = "RtApiPulse::probeDeviceOpen: error connecting output to PulseAudio server.";\r
-- goto error;\r
-- }\r
-- break;\r
-- default:\r
-- goto error;\r
-- }\r
--\r
-- if ( stream_.mode == UNINITIALIZED )\r
-- stream_.mode = mode;\r
-- else if ( stream_.mode == mode )\r
-- goto error;\r
-- else\r
-- stream_.mode = DUPLEX;\r
--\r
-- if ( !stream_.callbackInfo.isRunning ) {\r
-- stream_.callbackInfo.object = this;\r
-- stream_.callbackInfo.isRunning = true;\r
-- if ( pthread_create( &pah->thread, NULL, pulseaudio_callback, (void *)&stream_.callbackInfo) != 0 ) {\r
-- errorText_ = "RtApiPulse::probeDeviceOpen: error creating thread.";\r
-- goto error;\r
-- }\r
-- }\r
--\r
-- stream_.state = STREAM_STOPPED;\r
-- return true;\r
--\r
-- error:\r
-- if ( pah && stream_.callbackInfo.isRunning ) {\r
-- pthread_cond_destroy( &pah->runnable_cv );\r
-- delete pah;\r
-- stream_.apiHandle = 0;\r
-- }\r
--\r
-- for ( int i=0; i<2; i++ ) {\r
-- if ( stream_.userBuffer[i] ) {\r
-- free( stream_.userBuffer[i] );\r
-- stream_.userBuffer[i] = 0;\r
-- }\r
-- }\r
--\r
-- if ( stream_.deviceBuffer ) {\r
-- free( stream_.deviceBuffer );\r
-- stream_.deviceBuffer = 0;\r
-- }\r
--\r
-- return FAILURE;\r
--}\r
--\r
--//******************** End of __LINUX_PULSE__ *********************//\r
--#endif\r
--\r
--#if defined(__LINUX_OSS__)\r
--\r
--#include <unistd.h>\r
--#include <sys/ioctl.h>\r
--#include <unistd.h>\r
--#include <fcntl.h>\r
--#include <sys/soundcard.h>\r
--#include <errno.h>\r
--#include <math.h>\r
--\r
--static void *ossCallbackHandler(void * ptr);\r
--\r
--// A structure to hold various information related to the OSS API\r
--// implementation.\r
--struct OssHandle {\r
-- int id[2]; // device ids\r
-- bool xrun[2];\r
-- bool triggered;\r
-- pthread_cond_t runnable;\r
--\r
-- OssHandle()\r
-- :triggered(false) { id[0] = 0; id[1] = 0; xrun[0] = false; xrun[1] = false; }\r
--};\r
--\r
--RtApiOss :: RtApiOss()\r
--{\r
-- // Nothing to do here.\r
--}\r
--\r
--RtApiOss :: ~RtApiOss()\r
--{\r
-- if ( stream_.state != STREAM_CLOSED ) closeStream();\r
--}\r
--\r
--unsigned int RtApiOss :: getDeviceCount( void )\r
--{\r
-- int mixerfd = open( "/dev/mixer", O_RDWR, 0 );\r
-- if ( mixerfd == -1 ) {\r
-- errorText_ = "RtApiOss::getDeviceCount: error opening '/dev/mixer'.";\r
-- error( RtAudioError::WARNING );\r
-- return 0;\r
-- }\r
--\r
-- oss_sysinfo sysinfo;\r
-- if ( ioctl( mixerfd, SNDCTL_SYSINFO, &sysinfo ) == -1 ) {\r
-- close( mixerfd );\r
-- errorText_ = "RtApiOss::getDeviceCount: error getting sysinfo, OSS version >= 4.0 is required.";\r
-- error( RtAudioError::WARNING );\r
-- return 0;\r
-- }\r
--\r
-- close( mixerfd );\r
-- return sysinfo.numaudios;\r
--}\r
--\r
--RtAudio::DeviceInfo RtApiOss :: getDeviceInfo( unsigned int device )\r
--{\r
-- RtAudio::DeviceInfo info;\r
-- info.probed = false;\r
--\r
-- int mixerfd = open( "/dev/mixer", O_RDWR, 0 );\r
-- if ( mixerfd == -1 ) {\r
-- errorText_ = "RtApiOss::getDeviceInfo: error opening '/dev/mixer'.";\r
-- error( RtAudioError::WARNING );\r
-- return info;\r
-- }\r
--\r
-- oss_sysinfo sysinfo;\r
-- int result = ioctl( mixerfd, SNDCTL_SYSINFO, &sysinfo );\r
-- if ( result == -1 ) {\r
-- close( mixerfd );\r
-- errorText_ = "RtApiOss::getDeviceInfo: error getting sysinfo, OSS version >= 4.0 is required.";\r
-- error( RtAudioError::WARNING );\r
-- return info;\r
-- }\r
--\r
-- unsigned nDevices = sysinfo.numaudios;\r
-- if ( nDevices == 0 ) {\r
-- close( mixerfd );\r
-- errorText_ = "RtApiOss::getDeviceInfo: no devices found!";\r
-- error( RtAudioError::INVALID_USE );\r
-- return info;\r
-- }\r
--\r
-- if ( device >= nDevices ) {\r
-- close( mixerfd );\r
-- errorText_ = "RtApiOss::getDeviceInfo: device ID is invalid!";\r
-- error( RtAudioError::INVALID_USE );\r
-- return info;\r
-- }\r
--\r
-- oss_audioinfo ainfo;\r
-- ainfo.dev = device;\r
-- result = ioctl( mixerfd, SNDCTL_AUDIOINFO, &ainfo );\r
-- close( mixerfd );\r
-- if ( result == -1 ) {\r
-- errorStream_ << "RtApiOss::getDeviceInfo: error getting device (" << ainfo.name << ") info.";\r
-- errorText_ = errorStream_.str();\r
-- error( RtAudioError::WARNING );\r
-- return info;\r
-- }\r
--\r
-- // Probe channels\r
-- if ( ainfo.caps & PCM_CAP_OUTPUT ) info.outputChannels = ainfo.max_channels;\r
-- if ( ainfo.caps & PCM_CAP_INPUT ) info.inputChannels = ainfo.max_channels;\r
-- if ( ainfo.caps & PCM_CAP_DUPLEX ) {\r
-- if ( info.outputChannels > 0 && info.inputChannels > 0 && ainfo.caps & PCM_CAP_DUPLEX )\r
-- info.duplexChannels = (info.outputChannels > info.inputChannels) ? info.inputChannels : info.outputChannels;\r
-- }\r
--\r
-- // Probe data formats ... do for input\r
-- unsigned long mask = ainfo.iformats;\r
-- if ( mask & AFMT_S16_LE || mask & AFMT_S16_BE )\r
-- info.nativeFormats |= RTAUDIO_SINT16;\r
-- if ( mask & AFMT_S8 )\r
-- info.nativeFormats |= RTAUDIO_SINT8;\r
-- if ( mask & AFMT_S32_LE || mask & AFMT_S32_BE )\r
-- info.nativeFormats |= RTAUDIO_SINT32;\r
-- if ( mask & AFMT_FLOAT )\r
-- info.nativeFormats |= RTAUDIO_FLOAT32;\r
-- if ( mask & AFMT_S24_LE || mask & AFMT_S24_BE )\r
-- info.nativeFormats |= RTAUDIO_SINT24;\r
--\r
-- // Check that we have at least one supported format\r
-- if ( info.nativeFormats == 0 ) {\r
-- errorStream_ << "RtApiOss::getDeviceInfo: device (" << ainfo.name << ") data format not supported by RtAudio.";\r
-- errorText_ = errorStream_.str();\r
-- error( RtAudioError::WARNING );\r
-- return info;\r
-- }\r
--\r
-- // Probe the supported sample rates.\r
-- info.sampleRates.clear();\r
-- if ( ainfo.nrates ) {\r
-- for ( unsigned int i=0; i<ainfo.nrates; i++ ) {\r
-- for ( unsigned int k=0; k<MAX_SAMPLE_RATES; k++ ) {\r
-- if ( ainfo.rates[i] == SAMPLE_RATES[k] ) {\r
-- info.sampleRates.push_back( SAMPLE_RATES[k] );\r
--\r
-- if ( !info.preferredSampleRate || ( SAMPLE_RATES[k] <= 48000 && SAMPLE_RATES[k] > info.preferredSampleRate ) )\r
-- info.preferredSampleRate = SAMPLE_RATES[k];\r
--\r
-- break;\r
-- }\r
-- }\r
-- }\r
-- }\r
-- else {\r
-- // Check min and max rate values;\r
-- for ( unsigned int k=0; k<MAX_SAMPLE_RATES; k++ ) {\r
-- if ( ainfo.min_rate <= (int) SAMPLE_RATES[k] && ainfo.max_rate >= (int) SAMPLE_RATES[k] ) {\r
-- info.sampleRates.push_back( SAMPLE_RATES[k] );\r
--\r
-- if ( !info.preferredSampleRate || ( SAMPLE_RATES[k] <= 48000 && SAMPLE_RATES[k] > info.preferredSampleRate ) )\r
-- info.preferredSampleRate = SAMPLE_RATES[k];\r
-- }\r
-- }\r
-- }\r
--\r
-- if ( info.sampleRates.size() == 0 ) {\r
-- errorStream_ << "RtApiOss::getDeviceInfo: no supported sample rates found for device (" << ainfo.name << ").";\r
-- errorText_ = errorStream_.str();\r
-- error( RtAudioError::WARNING );\r
-- }\r
-- else {\r
-- info.probed = true;\r
-- info.name = ainfo.name;\r
-- }\r
--\r
-- return info;\r
--}\r
--\r
--\r
--bool RtApiOss :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,\r
-- unsigned int firstChannel, unsigned int sampleRate,\r
-- RtAudioFormat format, unsigned int *bufferSize,\r
-- RtAudio::StreamOptions *options )\r
--{\r
-- int mixerfd = open( "/dev/mixer", O_RDWR, 0 );\r
-- if ( mixerfd == -1 ) {\r
-- errorText_ = "RtApiOss::probeDeviceOpen: error opening '/dev/mixer'.";\r
-- return FAILURE;\r
-- }\r
--\r
-- oss_sysinfo sysinfo;\r
-- int result = ioctl( mixerfd, SNDCTL_SYSINFO, &sysinfo );\r
-- if ( result == -1 ) {\r
-- close( mixerfd );\r
-- errorText_ = "RtApiOss::probeDeviceOpen: error getting sysinfo, OSS version >= 4.0 is required.";\r
-- return FAILURE;\r
-- }\r
--\r
-- unsigned nDevices = sysinfo.numaudios;\r
-- if ( nDevices == 0 ) {\r
-- // This should not happen because a check is made before this function is called.\r
-- close( mixerfd );\r
-- errorText_ = "RtApiOss::probeDeviceOpen: no devices found!";\r
-- return FAILURE;\r
-- }\r
--\r
-- if ( device >= nDevices ) {\r
-- // This should not happen because a check is made before this function is called.\r
-- close( mixerfd );\r
-- errorText_ = "RtApiOss::probeDeviceOpen: device ID is invalid!";\r
-- return FAILURE;\r
-- }\r
--\r
-- oss_audioinfo ainfo;\r
-- ainfo.dev = device;\r
-- result = ioctl( mixerfd, SNDCTL_AUDIOINFO, &ainfo );\r
-- close( mixerfd );\r
-- if ( result == -1 ) {\r
-- errorStream_ << "RtApiOss::getDeviceInfo: error getting device (" << ainfo.name << ") info.";\r
-- errorText_ = errorStream_.str();\r
-- return FAILURE;\r
-- }\r
--\r
-- // Check if device supports input or output\r
-- if ( ( mode == OUTPUT && !( ainfo.caps & PCM_CAP_OUTPUT ) ) ||\r
-- ( mode == INPUT && !( ainfo.caps & PCM_CAP_INPUT ) ) ) {\r
-- if ( mode == OUTPUT )\r
-- errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") does not support output.";\r
-- else\r
-- errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") does not support input.";\r
-- errorText_ = errorStream_.str();\r
-- return FAILURE;\r
-- }\r
--\r
-- int flags = 0;\r
-- OssHandle *handle = (OssHandle *) stream_.apiHandle;\r
-- if ( mode == OUTPUT )\r
-- flags |= O_WRONLY;\r
-- else { // mode == INPUT\r
-- if (stream_.mode == OUTPUT && stream_.device[0] == device) {\r
-- // We just set the same device for playback ... close and reopen for duplex (OSS only).\r
-- close( handle->id[0] );\r
-- handle->id[0] = 0;\r
-- if ( !( ainfo.caps & PCM_CAP_DUPLEX ) ) {\r
-- errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") does not support duplex mode.";\r
-- errorText_ = errorStream_.str();\r
-- return FAILURE;\r
-- }\r
-- // Check that the number previously set channels is the same.\r
-- if ( stream_.nUserChannels[0] != channels ) {\r
-- errorStream_ << "RtApiOss::probeDeviceOpen: input/output channels must be equal for OSS duplex device (" << ainfo.name << ").";\r
-- errorText_ = errorStream_.str();\r
-- return FAILURE;\r
-- }\r
-- flags |= O_RDWR;\r
-- }\r
-- else\r
-- flags |= O_RDONLY;\r
-- }\r
--\r
-- // Set exclusive access if specified.\r
-- if ( options && options->flags & RTAUDIO_HOG_DEVICE ) flags |= O_EXCL;\r
--\r
-- // Try to open the device.\r
-- int fd;\r
-- fd = open( ainfo.devnode, flags, 0 );\r
-- if ( fd == -1 ) {\r
-- if ( errno == EBUSY )\r
-- errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") is busy.";\r
-- else\r
-- errorStream_ << "RtApiOss::probeDeviceOpen: error opening device (" << ainfo.name << ").";\r
-- errorText_ = errorStream_.str();\r
-- return FAILURE;\r
-- }\r
--\r
-- // For duplex operation, specifically set this mode (this doesn't seem to work).\r
-- /*\r
-- if ( flags | O_RDWR ) {\r
-- result = ioctl( fd, SNDCTL_DSP_SETDUPLEX, NULL );\r
-- if ( result == -1) {\r
-- errorStream_ << "RtApiOss::probeDeviceOpen: error setting duplex mode for device (" << ainfo.name << ").";\r
-- errorText_ = errorStream_.str();\r
-- return FAILURE;\r
-- }\r
-- }\r
-- */\r
--\r
-- // Check the device channel support.\r
-- stream_.nUserChannels[mode] = channels;\r
-- if ( ainfo.max_channels < (int)(channels + firstChannel) ) {\r
-- close( fd );\r
-- errorStream_ << "RtApiOss::probeDeviceOpen: the device (" << ainfo.name << ") does not support requested channel parameters.";\r
-- errorText_ = errorStream_.str();\r
-- return FAILURE;\r
-- }\r
--\r
-- // Set the number of channels.\r
-- int deviceChannels = channels + firstChannel;\r
-- result = ioctl( fd, SNDCTL_DSP_CHANNELS, &deviceChannels );\r
-- if ( result == -1 || deviceChannels < (int)(channels + firstChannel) ) {\r
-- close( fd );\r
-- errorStream_ << "RtApiOss::probeDeviceOpen: error setting channel parameters on device (" << ainfo.name << ").";\r
-- errorText_ = errorStream_.str();\r
-- return FAILURE;\r
-- }\r
-- stream_.nDeviceChannels[mode] = deviceChannels;\r
--\r
-- // Get the data format mask\r
-- int mask;\r
-- result = ioctl( fd, SNDCTL_DSP_GETFMTS, &mask );\r
-- if ( result == -1 ) {\r
-- close( fd );\r
-- errorStream_ << "RtApiOss::probeDeviceOpen: error getting device (" << ainfo.name << ") data formats.";\r
-- errorText_ = errorStream_.str();\r
-- return FAILURE;\r
-- }\r
--\r
-- // Determine how to set the device format.\r
-- stream_.userFormat = format;\r
-- int deviceFormat = -1;\r
-- stream_.doByteSwap[mode] = false;\r
-- if ( format == RTAUDIO_SINT8 ) {\r
-- if ( mask & AFMT_S8 ) {\r
-- deviceFormat = AFMT_S8;\r
-- stream_.deviceFormat[mode] = RTAUDIO_SINT8;\r
-- }\r
-- }\r
-- else if ( format == RTAUDIO_SINT16 ) {\r
-- if ( mask & AFMT_S16_NE ) {\r
-- deviceFormat = AFMT_S16_NE;\r
-- stream_.deviceFormat[mode] = RTAUDIO_SINT16;\r
-- }\r
-- else if ( mask & AFMT_S16_OE ) {\r
-- deviceFormat = AFMT_S16_OE;\r
-- stream_.deviceFormat[mode] = RTAUDIO_SINT16;\r
-- stream_.doByteSwap[mode] = true;\r
-- }\r
-- }\r
-- else if ( format == RTAUDIO_SINT24 ) {\r
-- if ( mask & AFMT_S24_NE ) {\r
-- deviceFormat = AFMT_S24_NE;\r
-- stream_.deviceFormat[mode] = RTAUDIO_SINT24;\r
-- }\r
-- else if ( mask & AFMT_S24_OE ) {\r
-- deviceFormat = AFMT_S24_OE;\r
-- stream_.deviceFormat[mode] = RTAUDIO_SINT24;\r
-- stream_.doByteSwap[mode] = true;\r
-- }\r
-- }\r
-- else if ( format == RTAUDIO_SINT32 ) {\r
-- if ( mask & AFMT_S32_NE ) {\r
-- deviceFormat = AFMT_S32_NE;\r
-- stream_.deviceFormat[mode] = RTAUDIO_SINT32;\r
-- }\r
-- else if ( mask & AFMT_S32_OE ) {\r
-- deviceFormat = AFMT_S32_OE;\r
-- stream_.deviceFormat[mode] = RTAUDIO_SINT32;\r
-- stream_.doByteSwap[mode] = true;\r
-- }\r
-- }\r
--\r
-- if ( deviceFormat == -1 ) {\r
-- // The user requested format is not natively supported by the device.\r
-- if ( mask & AFMT_S16_NE ) {\r
-- deviceFormat = AFMT_S16_NE;\r
-- stream_.deviceFormat[mode] = RTAUDIO_SINT16;\r
-- }\r
-- else if ( mask & AFMT_S32_NE ) {\r
-- deviceFormat = AFMT_S32_NE;\r
-- stream_.deviceFormat[mode] = RTAUDIO_SINT32;\r
-- }\r
-- else if ( mask & AFMT_S24_NE ) {\r
-- deviceFormat = AFMT_S24_NE;\r
-- stream_.deviceFormat[mode] = RTAUDIO_SINT24;\r
-- }\r
-- else if ( mask & AFMT_S16_OE ) {\r
-- deviceFormat = AFMT_S16_OE;\r
-- stream_.deviceFormat[mode] = RTAUDIO_SINT16;\r
-- stream_.doByteSwap[mode] = true;\r
-- }\r
-- else if ( mask & AFMT_S32_OE ) {\r
-- deviceFormat = AFMT_S32_OE;\r
-- stream_.deviceFormat[mode] = RTAUDIO_SINT32;\r
-- stream_.doByteSwap[mode] = true;\r
-- }\r
-- else if ( mask & AFMT_S24_OE ) {\r
-- deviceFormat = AFMT_S24_OE;\r
-- stream_.deviceFormat[mode] = RTAUDIO_SINT24;\r
-- stream_.doByteSwap[mode] = true;\r
-- }\r
-- else if ( mask & AFMT_S8) {\r
-- deviceFormat = AFMT_S8;\r
-- stream_.deviceFormat[mode] = RTAUDIO_SINT8;\r
-- }\r
-- }\r
--\r
-- if ( stream_.deviceFormat[mode] == 0 ) {\r
-- // This really shouldn't happen ...\r
-- close( fd );\r
-- errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") data format not supported by RtAudio.";\r
-- errorText_ = errorStream_.str();\r
-- return FAILURE;\r
-- }\r
--\r
-- // Set the data format.\r
-- int temp = deviceFormat;\r
-- result = ioctl( fd, SNDCTL_DSP_SETFMT, &deviceFormat );\r
-- if ( result == -1 || deviceFormat != temp ) {\r
-- close( fd );\r
-- errorStream_ << "RtApiOss::probeDeviceOpen: error setting data format on device (" << ainfo.name << ").";\r
-- errorText_ = errorStream_.str();\r
-- return FAILURE;\r
-- }\r
--\r
-- // Attempt to set the buffer size. According to OSS, the minimum\r
-- // number of buffers is two. The supposed minimum buffer size is 16\r
-- // bytes, so that will be our lower bound. The argument to this\r
-- // call is in the form 0xMMMMSSSS (hex), where the buffer size (in\r
-- // bytes) is given as 2^SSSS and the number of buffers as 2^MMMM.\r
-- // We'll check the actual value used near the end of the setup\r
-- // procedure.\r
-- int ossBufferBytes = *bufferSize * formatBytes( stream_.deviceFormat[mode] ) * deviceChannels;\r
-- if ( ossBufferBytes < 16 ) ossBufferBytes = 16;\r
-- int buffers = 0;\r
-- if ( options ) buffers = options->numberOfBuffers;\r
-- if ( options && options->flags & RTAUDIO_MINIMIZE_LATENCY ) buffers = 2;\r
-- if ( buffers < 2 ) buffers = 3;\r
-- temp = ((int) buffers << 16) + (int)( log10( (double)ossBufferBytes ) / log10( 2.0 ) );\r
-- result = ioctl( fd, SNDCTL_DSP_SETFRAGMENT, &temp );\r
-- if ( result == -1 ) {\r
-- close( fd );\r
-- errorStream_ << "RtApiOss::probeDeviceOpen: error setting buffer size on device (" << ainfo.name << ").";\r
-- errorText_ = errorStream_.str();\r
-- return FAILURE;\r
-- }\r
-- stream_.nBuffers = buffers;\r
--\r
-- // Save buffer size (in sample frames).\r
-- *bufferSize = ossBufferBytes / ( formatBytes(stream_.deviceFormat[mode]) * deviceChannels );\r
-- stream_.bufferSize = *bufferSize;\r
--\r
-- // Set the sample rate.\r
-- int srate = sampleRate;\r
-- result = ioctl( fd, SNDCTL_DSP_SPEED, &srate );\r
-- if ( result == -1 ) {\r
-- close( fd );\r
-- errorStream_ << "RtApiOss::probeDeviceOpen: error setting sample rate (" << sampleRate << ") on device (" << ainfo.name << ").";\r
-- errorText_ = errorStream_.str();\r
-- return FAILURE;\r
-- }\r
--\r
-- // Verify the sample rate setup worked.\r
-- if ( abs( srate - sampleRate ) > 100 ) {\r
-- close( fd );\r
-- errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") does not support sample rate (" << sampleRate << ").";\r
-- errorText_ = errorStream_.str();\r
-- return FAILURE;\r
-- }\r
-- stream_.sampleRate = sampleRate;\r
--\r
-- if ( mode == INPUT && stream_.mode == OUTPUT && stream_.device[0] == device) {\r
-- // We're doing duplex setup here.\r
-- stream_.deviceFormat[0] = stream_.deviceFormat[1];\r
-- stream_.nDeviceChannels[0] = deviceChannels;\r
-- }\r
--\r
-- // Set interleaving parameters.\r
-- stream_.userInterleaved = true;\r
-- stream_.deviceInterleaved[mode] = true;\r
-- if ( options && options->flags & RTAUDIO_NONINTERLEAVED )\r
-- stream_.userInterleaved = false;\r
--\r
-- // Set flags for buffer conversion\r
-- stream_.doConvertBuffer[mode] = false;\r
-- if ( stream_.userFormat != stream_.deviceFormat[mode] )\r
-- stream_.doConvertBuffer[mode] = true;\r
-- if ( stream_.nUserChannels[mode] < stream_.nDeviceChannels[mode] )\r
-- stream_.doConvertBuffer[mode] = true;\r
-- if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] &&\r
-- stream_.nUserChannels[mode] > 1 )\r
-- stream_.doConvertBuffer[mode] = true;\r
--\r
-- // Allocate the stream handles if necessary and then save.\r
-- if ( stream_.apiHandle == 0 ) {\r
-- try {\r
-- handle = new OssHandle;\r
-- }\r
-- catch ( std::bad_alloc& ) {\r
-- errorText_ = "RtApiOss::probeDeviceOpen: error allocating OssHandle memory.";\r
-- goto error;\r
-- }\r
--\r
-- if ( pthread_cond_init( &handle->runnable, NULL ) ) {\r
-- errorText_ = "RtApiOss::probeDeviceOpen: error initializing pthread condition variable.";\r
-- goto error;\r
-- }\r
--\r
-- stream_.apiHandle = (void *) handle;\r
-- }\r
-- else {\r
-- handle = (OssHandle *) stream_.apiHandle;\r
-- }\r
-- handle->id[mode] = fd;\r
--\r
-- // Allocate necessary internal buffers.\r
-- unsigned long bufferBytes;\r
-- bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat );\r
-- stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 );\r
-- if ( stream_.userBuffer[mode] == NULL ) {\r
-- errorText_ = "RtApiOss::probeDeviceOpen: error allocating user buffer memory.";\r
-- goto error;\r
-- }\r
--\r
-- if ( stream_.doConvertBuffer[mode] ) {\r
--\r
-- bool makeBuffer = true;\r
-- bufferBytes = stream_.nDeviceChannels[mode] * formatBytes( stream_.deviceFormat[mode] );\r
-- if ( mode == INPUT ) {\r
-- if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) {\r
-- unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] );\r
-- if ( bufferBytes <= bytesOut ) makeBuffer = false;\r
-- }\r
-- }\r
--\r
-- if ( makeBuffer ) {\r
-- bufferBytes *= *bufferSize;\r
-- if ( stream_.deviceBuffer ) free( stream_.deviceBuffer );\r
-- stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 );\r
-- if ( stream_.deviceBuffer == NULL ) {\r
-- errorText_ = "RtApiOss::probeDeviceOpen: error allocating device buffer memory.";\r
-- goto error;\r
-- }\r
-- }\r
-- }\r
--\r
-- stream_.device[mode] = device;\r
-- stream_.state = STREAM_STOPPED;\r
--\r
-- // Setup the buffer conversion information structure.\r
-- if ( stream_.doConvertBuffer[mode] ) setConvertInfo( mode, firstChannel );\r
--\r
-- // Setup thread if necessary.\r
-- if ( stream_.mode == OUTPUT && mode == INPUT ) {\r
-- // We had already set up an output stream.\r
-- stream_.mode = DUPLEX;\r
-- if ( stream_.device[0] == device ) handle->id[0] = fd;\r
-- }\r
-- else {\r
-- stream_.mode = mode;\r
--\r
-- // Setup callback thread.\r
-- stream_.callbackInfo.object = (void *) this;\r
--\r
-- // Set the thread attributes for joinable and realtime scheduling\r
-- // priority. The higher priority will only take affect if the\r
-- // program is run as root or suid.\r
-- pthread_attr_t attr;\r
-- pthread_attr_init( &attr );\r
-- pthread_attr_setdetachstate( &attr, PTHREAD_CREATE_JOINABLE );\r
--#ifdef SCHED_RR // Undefined with some OSes (eg: NetBSD 1.6.x with GNU Pthread)\r
-- if ( options && options->flags & RTAUDIO_SCHEDULE_REALTIME ) {\r
-- struct sched_param param;\r
-- int priority = options->priority;\r
-- int min = sched_get_priority_min( SCHED_RR );\r
-- int max = sched_get_priority_max( SCHED_RR );\r
-- if ( priority < min ) priority = min;\r
-- else if ( priority > max ) priority = max;\r
-- param.sched_priority = priority;\r
-- pthread_attr_setschedparam( &attr, ¶m );\r
-- pthread_attr_setschedpolicy( &attr, SCHED_RR );\r
-- }\r
-- else\r
-- pthread_attr_setschedpolicy( &attr, SCHED_OTHER );\r
--#else\r
-- pthread_attr_setschedpolicy( &attr, SCHED_OTHER );\r
--#endif\r
--\r
-- stream_.callbackInfo.isRunning = true;\r
-- result = pthread_create( &stream_.callbackInfo.thread, &attr, ossCallbackHandler, &stream_.callbackInfo );\r
-- pthread_attr_destroy( &attr );\r
-- if ( result ) {\r
-- stream_.callbackInfo.isRunning = false;\r
-- errorText_ = "RtApiOss::error creating callback thread!";\r
-- goto error;\r
-- }\r
-- }\r
--\r
-- return SUCCESS;\r
--\r
-- error:\r
-- if ( handle ) {\r
-- pthread_cond_destroy( &handle->runnable );\r
-- if ( handle->id[0] ) close( handle->id[0] );\r
-- if ( handle->id[1] ) close( handle->id[1] );\r
-- delete handle;\r
-- stream_.apiHandle = 0;\r
-- }\r
--\r
-- for ( int i=0; i<2; i++ ) {\r
-- if ( stream_.userBuffer[i] ) {\r
-- free( stream_.userBuffer[i] );\r
-- stream_.userBuffer[i] = 0;\r
-- }\r
-- }\r
--\r
-- if ( stream_.deviceBuffer ) {\r
-- free( stream_.deviceBuffer );\r
-- stream_.deviceBuffer = 0;\r
-- }\r
--\r
-- return FAILURE;\r
--}\r
--\r
--void RtApiOss :: closeStream()\r
--{\r
-- if ( stream_.state == STREAM_CLOSED ) {\r
-- errorText_ = "RtApiOss::closeStream(): no open stream to close!";\r
-- error( RtAudioError::WARNING );\r
-- return;\r
-- }\r
--\r
-- OssHandle *handle = (OssHandle *) stream_.apiHandle;\r
-- stream_.callbackInfo.isRunning = false;\r
-- MUTEX_LOCK( &stream_.mutex );\r
-- if ( stream_.state == STREAM_STOPPED )\r
-- pthread_cond_signal( &handle->runnable );\r
-- MUTEX_UNLOCK( &stream_.mutex );\r
-- pthread_join( stream_.callbackInfo.thread, NULL );\r
--\r
-- if ( stream_.state == STREAM_RUNNING ) {\r
-- if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX )\r
-- ioctl( handle->id[0], SNDCTL_DSP_HALT, 0 );\r
-- else\r
-- ioctl( handle->id[1], SNDCTL_DSP_HALT, 0 );\r
-- stream_.state = STREAM_STOPPED;\r
-- }\r
--\r
-- if ( handle ) {\r
-- pthread_cond_destroy( &handle->runnable );\r
-- if ( handle->id[0] ) close( handle->id[0] );\r
-- if ( handle->id[1] ) close( handle->id[1] );\r
-- delete handle;\r
-- stream_.apiHandle = 0;\r
-- }\r
--\r
-- for ( int i=0; i<2; i++ ) {\r
-- if ( stream_.userBuffer[i] ) {\r
-- free( stream_.userBuffer[i] );\r
-- stream_.userBuffer[i] = 0;\r
-- }\r
-- }\r
--\r
-- if ( stream_.deviceBuffer ) {\r
-- free( stream_.deviceBuffer );\r
-- stream_.deviceBuffer = 0;\r
-- }\r
--\r
-- stream_.mode = UNINITIALIZED;\r
-- stream_.state = STREAM_CLOSED;\r
--}\r
--\r
--void RtApiOss :: startStream()\r
--{\r
-- verifyStream();\r
-- if ( stream_.state == STREAM_RUNNING ) {\r
-- errorText_ = "RtApiOss::startStream(): the stream is already running!";\r
-- error( RtAudioError::WARNING );\r
-- return;\r
-- }\r
--\r
-- MUTEX_LOCK( &stream_.mutex );\r
--\r
-- stream_.state = STREAM_RUNNING;\r
--\r
-- // No need to do anything else here ... OSS automatically starts\r
-- // when fed samples.\r
--\r
-- MUTEX_UNLOCK( &stream_.mutex );\r
--\r
-- OssHandle *handle = (OssHandle *) stream_.apiHandle;\r
-- pthread_cond_signal( &handle->runnable );\r
--}\r
--\r
--void RtApiOss :: stopStream()\r
--{\r
-- verifyStream();\r
-- if ( stream_.state == STREAM_STOPPED ) {\r
-- errorText_ = "RtApiOss::stopStream(): the stream is already stopped!";\r
-- error( RtAudioError::WARNING );\r
-- return;\r
-- }\r
--\r
-- MUTEX_LOCK( &stream_.mutex );\r
--\r
-- // The state might change while waiting on a mutex.\r
-- if ( stream_.state == STREAM_STOPPED ) {\r
-- MUTEX_UNLOCK( &stream_.mutex );\r
-- return;\r
-- }\r
--\r
-- int result = 0;\r
-- OssHandle *handle = (OssHandle *) stream_.apiHandle;\r
-- if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {\r
--\r
-- // Flush the output with zeros a few times.\r
-- char *buffer;\r
-- int samples;\r
-- RtAudioFormat format;\r
--\r
-- if ( stream_.doConvertBuffer[0] ) {\r
-- buffer = stream_.deviceBuffer;\r
-- samples = stream_.bufferSize * stream_.nDeviceChannels[0];\r
-- format = stream_.deviceFormat[0];\r
-- }\r
-- else {\r
-- buffer = stream_.userBuffer[0];\r
-- samples = stream_.bufferSize * stream_.nUserChannels[0];\r
-- format = stream_.userFormat;\r
-- }\r
--\r
-- memset( buffer, 0, samples * formatBytes(format) );\r
-- for ( unsigned int i=0; i<stream_.nBuffers+1; i++ ) {\r
-- result = write( handle->id[0], buffer, samples * formatBytes(format) );\r
-- if ( result == -1 ) {\r
-- errorText_ = "RtApiOss::stopStream: audio write error.";\r
-- error( RtAudioError::WARNING );\r
-- }\r
-- }\r
--\r
-- result = ioctl( handle->id[0], SNDCTL_DSP_HALT, 0 );\r
-- if ( result == -1 ) {\r
-- errorStream_ << "RtApiOss::stopStream: system error stopping callback procedure on device (" << stream_.device[0] << ").";\r
-- errorText_ = errorStream_.str();\r
-- goto unlock;\r
-- }\r
-- handle->triggered = false;\r
-- }\r
--\r
-- if ( stream_.mode == INPUT || ( stream_.mode == DUPLEX && handle->id[0] != handle->id[1] ) ) {\r
-- result = ioctl( handle->id[1], SNDCTL_DSP_HALT, 0 );\r
-- if ( result == -1 ) {\r
-- errorStream_ << "RtApiOss::stopStream: system error stopping input callback procedure on device (" << stream_.device[0] << ").";\r
-- errorText_ = errorStream_.str();\r
-- goto unlock;\r
-- }\r
-- }\r
--\r
-- unlock:\r
-- stream_.state = STREAM_STOPPED;\r
-- MUTEX_UNLOCK( &stream_.mutex );\r
--\r
-- if ( result != -1 ) return;\r
-- error( RtAudioError::SYSTEM_ERROR );\r
--}\r
--\r
--void RtApiOss :: abortStream()\r
--{\r
-- verifyStream();\r
-- if ( stream_.state == STREAM_STOPPED ) {\r
-- errorText_ = "RtApiOss::abortStream(): the stream is already stopped!";\r
-- error( RtAudioError::WARNING );\r
-- return;\r
-- }\r
--\r
-- MUTEX_LOCK( &stream_.mutex );\r
--\r
-- // The state might change while waiting on a mutex.\r
-- if ( stream_.state == STREAM_STOPPED ) {\r
-- MUTEX_UNLOCK( &stream_.mutex );\r
-- return;\r
-- }\r
--\r
-- int result = 0;\r
-- OssHandle *handle = (OssHandle *) stream_.apiHandle;\r
-- if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {\r
-- result = ioctl( handle->id[0], SNDCTL_DSP_HALT, 0 );\r
-- if ( result == -1 ) {\r
-- errorStream_ << "RtApiOss::abortStream: system error stopping callback procedure on device (" << stream_.device[0] << ").";\r
-- errorText_ = errorStream_.str();\r
-- goto unlock;\r
-- }\r
-- handle->triggered = false;\r
-- }\r
--\r
-- if ( stream_.mode == INPUT || ( stream_.mode == DUPLEX && handle->id[0] != handle->id[1] ) ) {\r
-- result = ioctl( handle->id[1], SNDCTL_DSP_HALT, 0 );\r
-- if ( result == -1 ) {\r
-- errorStream_ << "RtApiOss::abortStream: system error stopping input callback procedure on device (" << stream_.device[0] << ").";\r
-- errorText_ = errorStream_.str();\r
-- goto unlock;\r
-- }\r
-- }\r
--\r
-- unlock:\r
-- stream_.state = STREAM_STOPPED;\r
-- MUTEX_UNLOCK( &stream_.mutex );\r
--\r
-- if ( result != -1 ) return;\r
-- error( RtAudioError::SYSTEM_ERROR );\r
--}\r
--\r
--void RtApiOss :: callbackEvent()\r
--{\r
-- OssHandle *handle = (OssHandle *) stream_.apiHandle;\r
-- if ( stream_.state == STREAM_STOPPED ) {\r
-- MUTEX_LOCK( &stream_.mutex );\r
-- pthread_cond_wait( &handle->runnable, &stream_.mutex );\r
-- if ( stream_.state != STREAM_RUNNING ) {\r
-- MUTEX_UNLOCK( &stream_.mutex );\r
-- return;\r
-- }\r
-- MUTEX_UNLOCK( &stream_.mutex );\r
-- }\r
--\r
-- if ( stream_.state == STREAM_CLOSED ) {\r
-- errorText_ = "RtApiOss::callbackEvent(): the stream is closed ... this shouldn't happen!";\r
-- error( RtAudioError::WARNING );\r
-- return;\r
-- }\r
--\r
-- // Invoke user callback to get fresh output data.\r
-- int doStopStream = 0;\r
-- RtAudioCallback callback = (RtAudioCallback) stream_.callbackInfo.callback;\r
-- double streamTime = getStreamTime();\r
-- RtAudioStreamStatus status = 0;\r
-- if ( stream_.mode != INPUT && handle->xrun[0] == true ) {\r
-- status |= RTAUDIO_OUTPUT_UNDERFLOW;\r
-- handle->xrun[0] = false;\r
-- }\r
-- if ( stream_.mode != OUTPUT && handle->xrun[1] == true ) {\r
-- status |= RTAUDIO_INPUT_OVERFLOW;\r
-- handle->xrun[1] = false;\r
-- }\r
-- doStopStream = callback( stream_.userBuffer[0], stream_.userBuffer[1],\r
-- stream_.bufferSize, streamTime, status, stream_.callbackInfo.userData );\r
-- if ( doStopStream == 2 ) {\r
-- this->abortStream();\r
-- return;\r
-- }\r
--\r
-- MUTEX_LOCK( &stream_.mutex );\r
--\r
-- // The state might change while waiting on a mutex.\r
-- if ( stream_.state == STREAM_STOPPED ) goto unlock;\r
--\r
-- int result;\r
-- char *buffer;\r
-- int samples;\r
-- RtAudioFormat format;\r
--\r
-- if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {\r
--\r
-- // Setup parameters and do buffer conversion if necessary.\r
-- if ( stream_.doConvertBuffer[0] ) {\r
-- buffer = stream_.deviceBuffer;\r
-- convertBuffer( buffer, stream_.userBuffer[0], stream_.convertInfo[0] );\r
-- samples = stream_.bufferSize * stream_.nDeviceChannels[0];\r
-- format = stream_.deviceFormat[0];\r
-- }\r
-- else {\r
-- buffer = stream_.userBuffer[0];\r
-- samples = stream_.bufferSize * stream_.nUserChannels[0];\r
-- format = stream_.userFormat;\r
-- }\r
--\r
-- // Do byte swapping if necessary.\r
-- if ( stream_.doByteSwap[0] )\r
-- byteSwapBuffer( buffer, samples, format );\r
--\r
-- if ( stream_.mode == DUPLEX && handle->triggered == false ) {\r
-- int trig = 0;\r
-- ioctl( handle->id[0], SNDCTL_DSP_SETTRIGGER, &trig );\r
-- result = write( handle->id[0], buffer, samples * formatBytes(format) );\r
-- trig = PCM_ENABLE_INPUT|PCM_ENABLE_OUTPUT;\r
-- ioctl( handle->id[0], SNDCTL_DSP_SETTRIGGER, &trig );\r
-- handle->triggered = true;\r
-- }\r
-- else\r
-- // Write samples to device.\r
-- result = write( handle->id[0], buffer, samples * formatBytes(format) );\r
--\r
-- if ( result == -1 ) {\r
-- // We'll assume this is an underrun, though there isn't a\r
-- // specific means for determining that.\r
-- handle->xrun[0] = true;\r
-- errorText_ = "RtApiOss::callbackEvent: audio write error.";\r
-- error( RtAudioError::WARNING );\r
-- // Continue on to input section.\r
-- }\r
-- }\r
--\r
-- if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {\r
--\r
-- // Setup parameters.\r
-- if ( stream_.doConvertBuffer[1] ) {\r
-- buffer = stream_.deviceBuffer;\r
-- samples = stream_.bufferSize * stream_.nDeviceChannels[1];\r
-- format = stream_.deviceFormat[1];\r
-- }\r
-- else {\r
-- buffer = stream_.userBuffer[1];\r
-- samples = stream_.bufferSize * stream_.nUserChannels[1];\r
-- format = stream_.userFormat;\r
-- }\r
--\r
-- // Read samples from device.\r
-- result = read( handle->id[1], buffer, samples * formatBytes(format) );\r
--\r
-- if ( result == -1 ) {\r
-- // We'll assume this is an overrun, though there isn't a\r
-- // specific means for determining that.\r
-- handle->xrun[1] = true;\r
-- errorText_ = "RtApiOss::callbackEvent: audio read error.";\r
-- error( RtAudioError::WARNING );\r
-- goto unlock;\r
-- }\r
--\r
-- // Do byte swapping if necessary.\r
-- if ( stream_.doByteSwap[1] )\r
-- byteSwapBuffer( buffer, samples, format );\r
--\r
-- // Do buffer conversion if necessary.\r
-- if ( stream_.doConvertBuffer[1] )\r
-- convertBuffer( stream_.userBuffer[1], stream_.deviceBuffer, stream_.convertInfo[1] );\r
-- }\r
--\r
-- unlock:\r
-- MUTEX_UNLOCK( &stream_.mutex );\r
--\r
-- RtApi::tickStreamTime();\r
-- if ( doStopStream == 1 ) this->stopStream();\r
--}\r
--\r
--static void *ossCallbackHandler( void *ptr )\r
--{\r
-- CallbackInfo *info = (CallbackInfo *) ptr;\r
-- RtApiOss *object = (RtApiOss *) info->object;\r
-- bool *isRunning = &info->isRunning;\r
--\r
-- while ( *isRunning == true ) {\r
-- pthread_testcancel();\r
-- object->callbackEvent();\r
-- }\r
--\r
-- pthread_exit( NULL );\r
--}\r
--\r
--//******************** End of __LINUX_OSS__ *********************//\r
--#endif\r
--\r
--\r
--// *************************************************** //\r
--//\r
--// Protected common (OS-independent) RtAudio methods.\r
--//\r
--// *************************************************** //\r
--\r
--// This method can be modified to control the behavior of error\r
--// message printing.\r
--void RtApi :: error( RtAudioError::Type type )\r
--{\r
-- errorStream_.str(""); // clear the ostringstream\r
--\r
-- RtAudioErrorCallback errorCallback = (RtAudioErrorCallback) stream_.callbackInfo.errorCallback;\r
-- if ( errorCallback ) {\r
-- // abortStream() can generate new error messages. Ignore them. Just keep original one.\r
--\r
-- if ( firstErrorOccurred_ )\r
-- return;\r
--\r
-- firstErrorOccurred_ = true;\r
-- const std::string errorMessage = errorText_;\r
--\r
-- if ( type != RtAudioError::WARNING && stream_.state != STREAM_STOPPED) {\r
-- stream_.callbackInfo.isRunning = false; // exit from the thread\r
-- abortStream();\r
-- }\r
--\r
-- errorCallback( type, errorMessage );\r
-- firstErrorOccurred_ = false;\r
-- return;\r
-- }\r
--\r
-- if ( type == RtAudioError::WARNING && showWarnings_ == true )\r
-- std::cerr << '\n' << errorText_ << "\n\n";\r
-- else if ( type != RtAudioError::WARNING )\r
-- throw( RtAudioError( errorText_, type ) );\r
--}\r
--\r
--void RtApi :: verifyStream()\r
--{\r
-- if ( stream_.state == STREAM_CLOSED ) {\r
-- errorText_ = "RtApi:: a stream is not open!";\r
-- error( RtAudioError::INVALID_USE );\r
-- }\r
--}\r
--\r
--void RtApi :: clearStreamInfo()\r
--{\r
-- stream_.mode = UNINITIALIZED;\r
-- stream_.state = STREAM_CLOSED;\r
-- stream_.sampleRate = 0;\r
-- stream_.bufferSize = 0;\r
-- stream_.nBuffers = 0;\r
-- stream_.userFormat = 0;\r
-- stream_.userInterleaved = true;\r
-- stream_.streamTime = 0.0;\r
-- stream_.apiHandle = 0;\r
-- stream_.deviceBuffer = 0;\r
-- stream_.callbackInfo.callback = 0;\r
-- stream_.callbackInfo.userData = 0;\r
-- stream_.callbackInfo.isRunning = false;\r
-- stream_.callbackInfo.errorCallback = 0;\r
-- for ( int i=0; i<2; i++ ) {\r
-- stream_.device[i] = 11111;\r
-- stream_.doConvertBuffer[i] = false;\r
-- stream_.deviceInterleaved[i] = true;\r
-- stream_.doByteSwap[i] = false;\r
-- stream_.nUserChannels[i] = 0;\r
-- stream_.nDeviceChannels[i] = 0;\r
-- stream_.channelOffset[i] = 0;\r
-- stream_.deviceFormat[i] = 0;\r
-- stream_.latency[i] = 0;\r
-- stream_.userBuffer[i] = 0;\r
-- stream_.convertInfo[i].channels = 0;\r
-- stream_.convertInfo[i].inJump = 0;\r
-- stream_.convertInfo[i].outJump = 0;\r
-- stream_.convertInfo[i].inFormat = 0;\r
-- stream_.convertInfo[i].outFormat = 0;\r
-- stream_.convertInfo[i].inOffset.clear();\r
-- stream_.convertInfo[i].outOffset.clear();\r
-- }\r
--}\r
--\r
--unsigned int RtApi :: formatBytes( RtAudioFormat format )\r
--{\r
-- if ( format == RTAUDIO_SINT16 )\r
-- return 2;\r
-- else if ( format == RTAUDIO_SINT32 || format == RTAUDIO_FLOAT32 )\r
-- return 4;\r
-- else if ( format == RTAUDIO_FLOAT64 )\r
-- return 8;\r
-- else if ( format == RTAUDIO_SINT24 )\r
-- return 3;\r
-- else if ( format == RTAUDIO_SINT8 )\r
-- return 1;\r
--\r
-- errorText_ = "RtApi::formatBytes: undefined format.";\r
-- error( RtAudioError::WARNING );\r
--\r
-- return 0;\r
--}\r
--\r
--void RtApi :: setConvertInfo( StreamMode mode, unsigned int firstChannel )\r
--{\r
-- if ( mode == INPUT ) { // convert device to user buffer\r
-- stream_.convertInfo[mode].inJump = stream_.nDeviceChannels[1];\r
-- stream_.convertInfo[mode].outJump = stream_.nUserChannels[1];\r
-- stream_.convertInfo[mode].inFormat = stream_.deviceFormat[1];\r
-- stream_.convertInfo[mode].outFormat = stream_.userFormat;\r
-- }\r
-- else { // convert user to device buffer\r
-- stream_.convertInfo[mode].inJump = stream_.nUserChannels[0];\r
-- stream_.convertInfo[mode].outJump = stream_.nDeviceChannels[0];\r
-- stream_.convertInfo[mode].inFormat = stream_.userFormat;\r
-- stream_.convertInfo[mode].outFormat = stream_.deviceFormat[0];\r
-- }\r
--\r
-- if ( stream_.convertInfo[mode].inJump < stream_.convertInfo[mode].outJump )\r
-- stream_.convertInfo[mode].channels = stream_.convertInfo[mode].inJump;\r
-- else\r
-- stream_.convertInfo[mode].channels = stream_.convertInfo[mode].outJump;\r
--\r
-- // Set up the interleave/deinterleave offsets.\r
-- if ( stream_.deviceInterleaved[mode] != stream_.userInterleaved ) {\r
-- if ( ( mode == OUTPUT && stream_.deviceInterleaved[mode] ) ||\r
-- ( mode == INPUT && stream_.userInterleaved ) ) {\r
-- for ( int k=0; k<stream_.convertInfo[mode].channels; k++ ) {\r
-- stream_.convertInfo[mode].inOffset.push_back( k * stream_.bufferSize );\r
-- stream_.convertInfo[mode].outOffset.push_back( k );\r
-- stream_.convertInfo[mode].inJump = 1;\r
-- }\r
-- }\r
-- else {\r
-- for ( int k=0; k<stream_.convertInfo[mode].channels; k++ ) {\r
-- stream_.convertInfo[mode].inOffset.push_back( k );\r
-- stream_.convertInfo[mode].outOffset.push_back( k * stream_.bufferSize );\r
-- stream_.convertInfo[mode].outJump = 1;\r
-- }\r
-- }\r
-- }\r
-- else { // no (de)interleaving\r
-- if ( stream_.userInterleaved ) {\r
-- for ( int k=0; k<stream_.convertInfo[mode].channels; k++ ) {\r
-- stream_.convertInfo[mode].inOffset.push_back( k );\r
-- stream_.convertInfo[mode].outOffset.push_back( k );\r
-- }\r
-- }\r
-- else {\r
-- for ( int k=0; k<stream_.convertInfo[mode].channels; k++ ) {\r
-- stream_.convertInfo[mode].inOffset.push_back( k * stream_.bufferSize );\r
-- stream_.convertInfo[mode].outOffset.push_back( k * stream_.bufferSize );\r
-- stream_.convertInfo[mode].inJump = 1;\r
-- stream_.convertInfo[mode].outJump = 1;\r
-- }\r
-- }\r
-- }\r
--\r
-- // Add channel offset.\r
-- if ( firstChannel > 0 ) {\r
-- if ( stream_.deviceInterleaved[mode] ) {\r
-- if ( mode == OUTPUT ) {\r
-- for ( int k=0; k<stream_.convertInfo[mode].channels; k++ )\r
-- stream_.convertInfo[mode].outOffset[k] += firstChannel;\r
-- }\r
-- else {\r
-- for ( int k=0; k<stream_.convertInfo[mode].channels; k++ )\r
-- stream_.convertInfo[mode].inOffset[k] += firstChannel;\r
-- }\r
-- }\r
-- else {\r
-- if ( mode == OUTPUT ) {\r
-- for ( int k=0; k<stream_.convertInfo[mode].channels; k++ )\r
-- stream_.convertInfo[mode].outOffset[k] += ( firstChannel * stream_.bufferSize );\r
-- }\r
-- else {\r
-- for ( int k=0; k<stream_.convertInfo[mode].channels; k++ )\r
-- stream_.convertInfo[mode].inOffset[k] += ( firstChannel * stream_.bufferSize );\r
-- }\r
-- }\r
-- }\r
--}\r
--\r
--void RtApi :: convertBuffer( char *outBuffer, char *inBuffer, ConvertInfo &info )\r
--{\r
-- // This function does format conversion, input/output channel compensation, and\r
-- // data interleaving/deinterleaving. 24-bit integers are assumed to occupy\r
-- // the lower three bytes of a 32-bit integer.\r
--\r
-- // Clear our device buffer when in/out duplex device channels are different\r
-- if ( outBuffer == stream_.deviceBuffer && stream_.mode == DUPLEX &&\r
-- ( stream_.nDeviceChannels[0] < stream_.nDeviceChannels[1] ) )\r
-- memset( outBuffer, 0, stream_.bufferSize * info.outJump * formatBytes( info.outFormat ) );\r
--\r
-- int j;\r
-- if (info.outFormat == RTAUDIO_FLOAT64) {\r
-- Float64 scale;\r
-- Float64 *out = (Float64 *)outBuffer;\r
--\r
-- if (info.inFormat == RTAUDIO_SINT8) {\r
-- signed char *in = (signed char *)inBuffer;\r
-- scale = 1.0 / 127.5;\r
-- for (unsigned int i=0; i<stream_.bufferSize; i++) {\r
-- for (j=0; j<info.channels; j++) {\r
-- out[info.outOffset[j]] = (Float64) in[info.inOffset[j]];\r
-- out[info.outOffset[j]] += 0.5;\r
-- out[info.outOffset[j]] *= scale;\r
-- }\r
-- in += info.inJump;\r
-- out += info.outJump;\r
-- }\r
-- }\r
-- else if (info.inFormat == RTAUDIO_SINT16) {\r
-- Int16 *in = (Int16 *)inBuffer;\r
-- scale = 1.0 / 32767.5;\r
-- for (unsigned int i=0; i<stream_.bufferSize; i++) {\r
-- for (j=0; j<info.channels; j++) {\r
-- out[info.outOffset[j]] = (Float64) in[info.inOffset[j]];\r
-- out[info.outOffset[j]] += 0.5;\r
-- out[info.outOffset[j]] *= scale;\r
-- }\r
-- in += info.inJump;\r
-- out += info.outJump;\r
-- }\r
-- }\r
-- else if (info.inFormat == RTAUDIO_SINT24) {\r
-- Int24 *in = (Int24 *)inBuffer;\r
-- scale = 1.0 / 8388607.5;\r
-- for (unsigned int i=0; i<stream_.bufferSize; i++) {\r
-- for (j=0; j<info.channels; j++) {\r
-- out[info.outOffset[j]] = (Float64) (in[info.inOffset[j]].asInt());\r
-- out[info.outOffset[j]] += 0.5;\r
-- out[info.outOffset[j]] *= scale;\r
-- }\r
-- in += info.inJump;\r
-- out += info.outJump;\r
-- }\r
-- }\r
-- else if (info.inFormat == RTAUDIO_SINT32) {\r
-- Int32 *in = (Int32 *)inBuffer;\r
-- scale = 1.0 / 2147483647.5;\r
-- for (unsigned int i=0; i<stream_.bufferSize; i++) {\r
-- for (j=0; j<info.channels; j++) {\r
-- out[info.outOffset[j]] = (Float64) in[info.inOffset[j]];\r
-- out[info.outOffset[j]] += 0.5;\r
-- out[info.outOffset[j]] *= scale;\r
-- }\r
-- in += info.inJump;\r
-- out += info.outJump;\r
-- }\r
-- }\r
-- else if (info.inFormat == RTAUDIO_FLOAT32) {\r
-- Float32 *in = (Float32 *)inBuffer;\r
-- for (unsigned int i=0; i<stream_.bufferSize; i++) {\r
-- for (j=0; j<info.channels; j++) {\r
-- out[info.outOffset[j]] = (Float64) in[info.inOffset[j]];\r
-- }\r
-- in += info.inJump;\r
-- out += info.outJump;\r
-- }\r
-- }\r
-- else if (info.inFormat == RTAUDIO_FLOAT64) {\r
-- // Channel compensation and/or (de)interleaving only.\r
-- Float64 *in = (Float64 *)inBuffer;\r
-- for (unsigned int i=0; i<stream_.bufferSize; i++) {\r
-- for (j=0; j<info.channels; j++) {\r
-- out[info.outOffset[j]] = in[info.inOffset[j]];\r
-- }\r
-- in += info.inJump;\r
-- out += info.outJump;\r
-- }\r
-- }\r
-- }\r
-- else if (info.outFormat == RTAUDIO_FLOAT32) {\r
-- Float32 scale;\r
-- Float32 *out = (Float32 *)outBuffer;\r
--\r
-- if (info.inFormat == RTAUDIO_SINT8) {\r
-- signed char *in = (signed char *)inBuffer;\r
-- scale = (Float32) ( 1.0 / 127.5 );\r
-- for (unsigned int i=0; i<stream_.bufferSize; i++) {\r
-- for (j=0; j<info.channels; j++) {\r
-- out[info.outOffset[j]] = (Float32) in[info.inOffset[j]];\r
-- out[info.outOffset[j]] += 0.5;\r
-- out[info.outOffset[j]] *= scale;\r
-- }\r
-- in += info.inJump;\r
-- out += info.outJump;\r
-- }\r
-- }\r
-- else if (info.inFormat == RTAUDIO_SINT16) {\r
-- Int16 *in = (Int16 *)inBuffer;\r
-- scale = (Float32) ( 1.0 / 32767.5 );\r
-- for (unsigned int i=0; i<stream_.bufferSize; i++) {\r
-- for (j=0; j<info.channels; j++) {\r
-- out[info.outOffset[j]] = (Float32) in[info.inOffset[j]];\r
-- out[info.outOffset[j]] += 0.5;\r
-- out[info.outOffset[j]] *= scale;\r
-- }\r
-- in += info.inJump;\r
-- out += info.outJump;\r
-- }\r
-- }\r
-- else if (info.inFormat == RTAUDIO_SINT24) {\r
-- Int24 *in = (Int24 *)inBuffer;\r
-- scale = (Float32) ( 1.0 / 8388607.5 );\r
-- for (unsigned int i=0; i<stream_.bufferSize; i++) {\r
-- for (j=0; j<info.channels; j++) {\r
-- out[info.outOffset[j]] = (Float32) (in[info.inOffset[j]].asInt());\r
-- out[info.outOffset[j]] += 0.5;\r
-- out[info.outOffset[j]] *= scale;\r
-- }\r
-- in += info.inJump;\r
-- out += info.outJump;\r
-- }\r
-- }\r
-- else if (info.inFormat == RTAUDIO_SINT32) {\r
-- Int32 *in = (Int32 *)inBuffer;\r
-- scale = (Float32) ( 1.0 / 2147483647.5 );\r
-- for (unsigned int i=0; i<stream_.bufferSize; i++) {\r
-- for (j=0; j<info.channels; j++) {\r
-- out[info.outOffset[j]] = (Float32) in[info.inOffset[j]];\r
-- out[info.outOffset[j]] += 0.5;\r
-- out[info.outOffset[j]] *= scale;\r
-- }\r
-- in += info.inJump;\r
-- out += info.outJump;\r
-- }\r
-- }\r
-- else if (info.inFormat == RTAUDIO_FLOAT32) {\r
-- // Channel compensation and/or (de)interleaving only.\r
-- Float32 *in = (Float32 *)inBuffer;\r
-- for (unsigned int i=0; i<stream_.bufferSize; i++) {\r
-- for (j=0; j<info.channels; j++) {\r
-- out[info.outOffset[j]] = in[info.inOffset[j]];\r
-- }\r
-- in += info.inJump;\r
-- out += info.outJump;\r
-- }\r
-- }\r
-- else if (info.inFormat == RTAUDIO_FLOAT64) {\r
-- Float64 *in = (Float64 *)inBuffer;\r
-- for (unsigned int i=0; i<stream_.bufferSize; i++) {\r
-- for (j=0; j<info.channels; j++) {\r
-- out[info.outOffset[j]] = (Float32) in[info.inOffset[j]];\r
-- }\r
-- in += info.inJump;\r
-- out += info.outJump;\r
-- }\r
-- }\r
-- }\r
-- else if (info.outFormat == RTAUDIO_SINT32) {\r
-- Int32 *out = (Int32 *)outBuffer;\r
-- if (info.inFormat == RTAUDIO_SINT8) {\r
-- signed char *in = (signed char *)inBuffer;\r
-- for (unsigned int i=0; i<stream_.bufferSize; i++) {\r
-- for (j=0; j<info.channels; j++) {\r
-- out[info.outOffset[j]] = (Int32) in[info.inOffset[j]];\r
-- out[info.outOffset[j]] <<= 24;\r
-- }\r
-- in += info.inJump;\r
-- out += info.outJump;\r
-- }\r
-- }\r
-- else if (info.inFormat == RTAUDIO_SINT16) {\r
-- Int16 *in = (Int16 *)inBuffer;\r
-- for (unsigned int i=0; i<stream_.bufferSize; i++) {\r
-- for (j=0; j<info.channels; j++) {\r
-- out[info.outOffset[j]] = (Int32) in[info.inOffset[j]];\r
-- out[info.outOffset[j]] <<= 16;\r
-- }\r
-- in += info.inJump;\r
-- out += info.outJump;\r
-- }\r
-- }\r
-- else if (info.inFormat == RTAUDIO_SINT24) {\r
-- Int24 *in = (Int24 *)inBuffer;\r
-- for (unsigned int i=0; i<stream_.bufferSize; i++) {\r
-- for (j=0; j<info.channels; j++) {\r
-- out[info.outOffset[j]] = (Int32) in[info.inOffset[j]].asInt();\r
-- out[info.outOffset[j]] <<= 8;\r
-- }\r
-- in += info.inJump;\r
-- out += info.outJump;\r
-- }\r
-- }\r
-- else if (info.inFormat == RTAUDIO_SINT32) {\r
-- // Channel compensation and/or (de)interleaving only.\r
-- Int32 *in = (Int32 *)inBuffer;\r
-- for (unsigned int i=0; i<stream_.bufferSize; i++) {\r
-- for (j=0; j<info.channels; j++) {\r
-- out[info.outOffset[j]] = in[info.inOffset[j]];\r
-- }\r
-- in += info.inJump;\r
-- out += info.outJump;\r
-- }\r
-- }\r
-- else if (info.inFormat == RTAUDIO_FLOAT32) {\r
-- Float32 *in = (Float32 *)inBuffer;\r
-- for (unsigned int i=0; i<stream_.bufferSize; i++) {\r
-- for (j=0; j<info.channels; j++) {\r
-- out[info.outOffset[j]] = (Int32) (in[info.inOffset[j]] * 2147483647.5 - 0.5);\r
-- }\r
-- in += info.inJump;\r
-- out += info.outJump;\r
-- }\r
-- }\r
-- else if (info.inFormat == RTAUDIO_FLOAT64) {\r
-- Float64 *in = (Float64 *)inBuffer;\r
-- for (unsigned int i=0; i<stream_.bufferSize; i++) {\r
-- for (j=0; j<info.channels; j++) {\r
-- out[info.outOffset[j]] = (Int32) (in[info.inOffset[j]] * 2147483647.5 - 0.5);\r
-- }\r
-- in += info.inJump;\r
-- out += info.outJump;\r
-- }\r
-- }\r
-- }\r
-- else if (info.outFormat == RTAUDIO_SINT24) {\r
-- Int24 *out = (Int24 *)outBuffer;\r
-- if (info.inFormat == RTAUDIO_SINT8) {\r
-- signed char *in = (signed char *)inBuffer;\r
-- for (unsigned int i=0; i<stream_.bufferSize; i++) {\r
-- for (j=0; j<info.channels; j++) {\r
-- out[info.outOffset[j]] = (Int32) (in[info.inOffset[j]] << 16);\r
-- //out[info.outOffset[j]] <<= 16;\r
-- }\r
-- in += info.inJump;\r
-- out += info.outJump;\r
-- }\r
-- }\r
-- else if (info.inFormat == RTAUDIO_SINT16) {\r
-- Int16 *in = (Int16 *)inBuffer;\r
-- for (unsigned int i=0; i<stream_.bufferSize; i++) {\r
-- for (j=0; j<info.channels; j++) {\r
-- out[info.outOffset[j]] = (Int32) (in[info.inOffset[j]] << 8);\r
-- //out[info.outOffset[j]] <<= 8;\r
-- }\r
-- in += info.inJump;\r
-- out += info.outJump;\r
-- }\r
-- }\r
-- else if (info.inFormat == RTAUDIO_SINT24) {\r
-- // Channel compensation and/or (de)interleaving only.\r
-- Int24 *in = (Int24 *)inBuffer;\r
-- for (unsigned int i=0; i<stream_.bufferSize; i++) {\r
-- for (j=0; j<info.channels; j++) {\r
-- out[info.outOffset[j]] = in[info.inOffset[j]];\r
-- }\r
-- in += info.inJump;\r
-- out += info.outJump;\r
-- }\r
-- }\r
-- else if (info.inFormat == RTAUDIO_SINT32) {\r
-- Int32 *in = (Int32 *)inBuffer;\r
-- for (unsigned int i=0; i<stream_.bufferSize; i++) {\r
-- for (j=0; j<info.channels; j++) {\r
-- out[info.outOffset[j]] = (Int32) (in[info.inOffset[j]] >> 8);\r
-- //out[info.outOffset[j]] >>= 8;\r
-- }\r
-- in += info.inJump;\r
-- out += info.outJump;\r
-- }\r
-- }\r
-- else if (info.inFormat == RTAUDIO_FLOAT32) {\r
-- Float32 *in = (Float32 *)inBuffer;\r
-- for (unsigned int i=0; i<stream_.bufferSize; i++) {\r
-- for (j=0; j<info.channels; j++) {\r
-- out[info.outOffset[j]] = (Int32) (in[info.inOffset[j]] * 8388607.5 - 0.5);\r
-- }\r
-- in += info.inJump;\r
-- out += info.outJump;\r
-- }\r
-- }\r
-- else if (info.inFormat == RTAUDIO_FLOAT64) {\r
-- Float64 *in = (Float64 *)inBuffer;\r
-- for (unsigned int i=0; i<stream_.bufferSize; i++) {\r
-- for (j=0; j<info.channels; j++) {\r
-- out[info.outOffset[j]] = (Int32) (in[info.inOffset[j]] * 8388607.5 - 0.5);\r
-- }\r
-- in += info.inJump;\r
-- out += info.outJump;\r
-- }\r
-- }\r
-- }\r
-- else if (info.outFormat == RTAUDIO_SINT16) {\r
-- Int16 *out = (Int16 *)outBuffer;\r
-- if (info.inFormat == RTAUDIO_SINT8) {\r
-- signed char *in = (signed char *)inBuffer;\r
-- for (unsigned int i=0; i<stream_.bufferSize; i++) {\r
-- for (j=0; j<info.channels; j++) {\r
-- out[info.outOffset[j]] = (Int16) in[info.inOffset[j]];\r
-- out[info.outOffset[j]] <<= 8;\r
-- }\r
-- in += info.inJump;\r
-- out += info.outJump;\r
-- }\r
-- }\r
-- else if (info.inFormat == RTAUDIO_SINT16) {\r
-- // Channel compensation and/or (de)interleaving only.\r
-- Int16 *in = (Int16 *)inBuffer;\r
-- for (unsigned int i=0; i<stream_.bufferSize; i++) {\r
-- for (j=0; j<info.channels; j++) {\r
-- out[info.outOffset[j]] = in[info.inOffset[j]];\r
-- }\r
-- in += info.inJump;\r
-- out += info.outJump;\r
-- }\r
-- }\r
-- else if (info.inFormat == RTAUDIO_SINT24) {\r
-- Int24 *in = (Int24 *)inBuffer;\r
-- for (unsigned int i=0; i<stream_.bufferSize; i++) {\r
-- for (j=0; j<info.channels; j++) {\r
-- out[info.outOffset[j]] = (Int16) (in[info.inOffset[j]].asInt() >> 8);\r
-- }\r
-- in += info.inJump;\r
-- out += info.outJump;\r
-- }\r
-- }\r
-- else if (info.inFormat == RTAUDIO_SINT32) {\r
-- Int32 *in = (Int32 *)inBuffer;\r
-- for (unsigned int i=0; i<stream_.bufferSize; i++) {\r
-- for (j=0; j<info.channels; j++) {\r
-- out[info.outOffset[j]] = (Int16) ((in[info.inOffset[j]] >> 16) & 0x0000ffff);\r
-- }\r
-- in += info.inJump;\r
-- out += info.outJump;\r
-- }\r
-- }\r
-- else if (info.inFormat == RTAUDIO_FLOAT32) {\r
-- Float32 *in = (Float32 *)inBuffer;\r
-- for (unsigned int i=0; i<stream_.bufferSize; i++) {\r
-- for (j=0; j<info.channels; j++) {\r
-- out[info.outOffset[j]] = (Int16) (in[info.inOffset[j]] * 32767.5 - 0.5);\r
-- }\r
-- in += info.inJump;\r
-- out += info.outJump;\r
-- }\r
-- }\r
-- else if (info.inFormat == RTAUDIO_FLOAT64) {\r
-- Float64 *in = (Float64 *)inBuffer;\r
-- for (unsigned int i=0; i<stream_.bufferSize; i++) {\r
-- for (j=0; j<info.channels; j++) {\r
-- out[info.outOffset[j]] = (Int16) (in[info.inOffset[j]] * 32767.5 - 0.5);\r
-- }\r
-- in += info.inJump;\r
-- out += info.outJump;\r
-- }\r
-- }\r
-- }\r
-- else if (info.outFormat == RTAUDIO_SINT8) {\r
-- signed char *out = (signed char *)outBuffer;\r
-- if (info.inFormat == RTAUDIO_SINT8) {\r
-- // Channel compensation and/or (de)interleaving only.\r
-- signed char *in = (signed char *)inBuffer;\r
-- for (unsigned int i=0; i<stream_.bufferSize; i++) {\r
-- for (j=0; j<info.channels; j++) {\r
-- out[info.outOffset[j]] = in[info.inOffset[j]];\r
-- }\r
-- in += info.inJump;\r
-- out += info.outJump;\r
-- }\r
-- }\r
-- if (info.inFormat == RTAUDIO_SINT16) {\r
-- Int16 *in = (Int16 *)inBuffer;\r
-- for (unsigned int i=0; i<stream_.bufferSize; i++) {\r
-- for (j=0; j<info.channels; j++) {\r
-- out[info.outOffset[j]] = (signed char) ((in[info.inOffset[j]] >> 8) & 0x00ff);\r
-- }\r
-- in += info.inJump;\r
-- out += info.outJump;\r
-- }\r
-- }\r
-- else if (info.inFormat == RTAUDIO_SINT24) {\r
-- Int24 *in = (Int24 *)inBuffer;\r
-- for (unsigned int i=0; i<stream_.bufferSize; i++) {\r
-- for (j=0; j<info.channels; j++) {\r
-- out[info.outOffset[j]] = (signed char) (in[info.inOffset[j]].asInt() >> 16);\r
-- }\r
-- in += info.inJump;\r
-- out += info.outJump;\r
-- }\r
-- }\r
-- else if (info.inFormat == RTAUDIO_SINT32) {\r
-- Int32 *in = (Int32 *)inBuffer;\r
-- for (unsigned int i=0; i<stream_.bufferSize; i++) {\r
-- for (j=0; j<info.channels; j++) {\r
-- out[info.outOffset[j]] = (signed char) ((in[info.inOffset[j]] >> 24) & 0x000000ff);\r
-- }\r
-- in += info.inJump;\r
-- out += info.outJump;\r
-- }\r
-- }\r
-- else if (info.inFormat == RTAUDIO_FLOAT32) {\r
-- Float32 *in = (Float32 *)inBuffer;\r
-- for (unsigned int i=0; i<stream_.bufferSize; i++) {\r
-- for (j=0; j<info.channels; j++) {\r
-- out[info.outOffset[j]] = (signed char) (in[info.inOffset[j]] * 127.5 - 0.5);\r
-- }\r
-- in += info.inJump;\r
-- out += info.outJump;\r
-- }\r
-- }\r
-- else if (info.inFormat == RTAUDIO_FLOAT64) {\r
-- Float64 *in = (Float64 *)inBuffer;\r
-- for (unsigned int i=0; i<stream_.bufferSize; i++) {\r
-- for (j=0; j<info.channels; j++) {\r
-- out[info.outOffset[j]] = (signed char) (in[info.inOffset[j]] * 127.5 - 0.5);\r
-- }\r
-- in += info.inJump;\r
-- out += info.outJump;\r
-- }\r
-- }\r
-- }\r
--}\r
--\r
--//static inline uint16_t bswap_16(uint16_t x) { return (x>>8) | (x<<8); }\r
--//static inline uint32_t bswap_32(uint32_t x) { return (bswap_16(x&0xffff)<<16) | (bswap_16(x>>16)); }\r
--//static inline uint64_t bswap_64(uint64_t x) { return (((unsigned long long)bswap_32(x&0xffffffffull))<<32) | (bswap_32(x>>32)); }\r
--\r
--void RtApi :: byteSwapBuffer( char *buffer, unsigned int samples, RtAudioFormat format )\r
--{\r
-- char val;\r
-- char *ptr;\r
--\r
-- ptr = buffer;\r
-- if ( format == RTAUDIO_SINT16 ) {\r
-- for ( unsigned int i=0; i<samples; i++ ) {\r
-- // Swap 1st and 2nd bytes.\r
-- val = *(ptr);\r
-- *(ptr) = *(ptr+1);\r
-- *(ptr+1) = val;\r
--\r
-- // Increment 2 bytes.\r
-- ptr += 2;\r
-- }\r
-- }\r
-- else if ( format == RTAUDIO_SINT32 ||\r
-- format == RTAUDIO_FLOAT32 ) {\r
-- for ( unsigned int i=0; i<samples; i++ ) {\r
-- // Swap 1st and 4th bytes.\r
-- val = *(ptr);\r
-- *(ptr) = *(ptr+3);\r
-- *(ptr+3) = val;\r
--\r
-- // Swap 2nd and 3rd bytes.\r
-- ptr += 1;\r
-- val = *(ptr);\r
-- *(ptr) = *(ptr+1);\r
-- *(ptr+1) = val;\r
--\r
-- // Increment 3 more bytes.\r
-- ptr += 3;\r
-- }\r
-- }\r
-- else if ( format == RTAUDIO_SINT24 ) {\r
-- for ( unsigned int i=0; i<samples; i++ ) {\r
-- // Swap 1st and 3rd bytes.\r
-- val = *(ptr);\r
-- *(ptr) = *(ptr+2);\r
-- *(ptr+2) = val;\r
--\r
-- // Increment 2 more bytes.\r
-- ptr += 2;\r
-- }\r
-- }\r
-- else if ( format == RTAUDIO_FLOAT64 ) {\r
-- for ( unsigned int i=0; i<samples; i++ ) {\r
-- // Swap 1st and 8th bytes\r
-- val = *(ptr);\r
-- *(ptr) = *(ptr+7);\r
-- *(ptr+7) = val;\r
--\r
-- // Swap 2nd and 7th bytes\r
-- ptr += 1;\r
-- val = *(ptr);\r
-- *(ptr) = *(ptr+5);\r
-- *(ptr+5) = val;\r
--\r
-- // Swap 3rd and 6th bytes\r
-- ptr += 1;\r
-- val = *(ptr);\r
-- *(ptr) = *(ptr+3);\r
-- *(ptr+3) = val;\r
--\r
-- // Swap 4th and 5th bytes\r
-- ptr += 1;\r
-- val = *(ptr);\r
-- *(ptr) = *(ptr+1);\r
-- *(ptr+1) = val;\r
--\r
-- // Increment 5 more bytes.\r
-- ptr += 5;\r
-- }\r
-- }\r
--}\r
--\r
-- // Indentation settings for Vim and Emacs\r
-- //\r
-- // Local Variables:\r
-- // c-basic-offset: 2\r
-- // indent-tabs-mode: nil\r
-- // End:\r
-- //\r
-- // vim: et sts=2 sw=2\r
-+/************************************************************************/
-+/*! \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;
-+}
-+
-+void RtAudio :: getCompiledApi( std::vector<RtAudio::Api> &apis )
-+{
-+ apis.clear();
-+
-+ // The order here will control the order of RtAudio's API search in
-+ // the constructor.
-+#if defined(__UNIX_JACK__)
-+ apis.push_back( UNIX_JACK );
-+#endif
-+#if defined(__LINUX_ALSA__)
-+ apis.push_back( LINUX_ALSA );
-+#endif
-+#if defined(__LINUX_PULSE__)
-+ apis.push_back( LINUX_PULSE );
-+#endif
-+#if defined(__LINUX_OSS__)
-+ apis.push_back( LINUX_OSS );
-+#endif
-+#if defined(__WINDOWS_ASIO__)
-+ apis.push_back( WINDOWS_ASIO );
-+#endif
-+#if defined(__WINDOWS_WASAPI__)
-+ apis.push_back( WINDOWS_WASAPI );
-+#endif
-+#if defined(__WINDOWS_DS__)
-+ apis.push_back( WINDOWS_DS );
-+#endif
-+#if defined(__MACOSX_CORE__)
-+ apis.push_back( MACOSX_CORE );
-+#endif
-+#if defined(__RTAUDIO_DUMMY__)
-+ apis.push_back( RTAUDIO_DUMMY );
-+#endif
-+}
-+
-+void RtAudio :: openRtApi( RtAudio::Api api )
-+{
-+ if ( rtapi_ )
-+ delete rtapi_;
-+ rtapi_ = 0;
-+
-+#if defined(__UNIX_JACK__)
-+ if ( api == UNIX_JACK )
-+ rtapi_ = new RtApiJack();
-+#endif
-+#if defined(__LINUX_ALSA__)
-+ if ( api == LINUX_ALSA )
-+ rtapi_ = new RtApiAlsa();
-+#endif
-+#if defined(__LINUX_PULSE__)
-+ if ( api == LINUX_PULSE )
-+ rtapi_ = new RtApiPulse();
-+#endif
-+#if defined(__LINUX_OSS__)
-+ if ( api == LINUX_OSS )
-+ rtapi_ = new RtApiOss();
-+#endif
-+#if defined(__WINDOWS_ASIO__)
-+ if ( api == WINDOWS_ASIO )
-+ rtapi_ = new RtApiAsio();
-+#endif
-+#if defined(__WINDOWS_WASAPI__)
-+ if ( api == WINDOWS_WASAPI )
-+ rtapi_ = new RtApiWasapi();
-+#endif
-+#if defined(__WINDOWS_DS__)
-+ if ( api == WINDOWS_DS )
-+ rtapi_ = new RtApiDs();
-+#endif
-+#if defined(__MACOSX_CORE__)
-+ if ( api == MACOSX_CORE )
-+ rtapi_ = new RtApiCore();
-+#endif
-+#if defined(__RTAUDIO_DUMMY__)
-+ if ( api == RTAUDIO_DUMMY )
-+ rtapi_ = new RtApiDummy();
-+#endif
-+}
-+
-+RtAudio :: RtAudio( RtAudio::Api api )
-+{
-+ rtapi_ = 0;
-+
-+ if ( api != UNSPECIFIED ) {
-+ // Attempt to open the specified API.
-+ openRtApi( api );
-+ if ( rtapi_ ) return;
-+
-+ // No compiled support for specified API value. Issue a debug
-+ // warning and continue as if no API was specified.
-+ std::cerr << "\nRtAudio: no compiled support for specified API argument!\n" << std::endl;
-+ }
-+
-+ // Iterate through the compiled APIs and return as soon as we find
-+ // one with at least one device or we reach the end of the list.
-+ std::vector< RtAudio::Api > apis;
-+ getCompiledApi( apis );
-+ for ( unsigned int i=0; i<apis.size(); i++ ) {
-+ openRtApi( apis[i] );
-+ if ( rtapi_ && rtapi_->getDeviceCount() ) break;
-+ }
-+
-+ if ( rtapi_ ) return;
-+
-+ // It should not be possible to get here because the preprocessor
-+ // definition __RTAUDIO_DUMMY__ is automatically defined if no
-+ // API-specific definitions are passed to the compiler. But just in
-+ // case something weird happens, we'll thow an error.
-+ std::string errorText = "\nRtAudio: no compiled API support found ... critical error!!\n\n";
-+ throw( RtAudioError( errorText, RtAudioError::UNSPECIFIED ) );
-+}
-+
-+RtAudio :: ~RtAudio()
-+{
-+ 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, NULL, 0 );
-+ if ( ports ) {
-+ // Parse the port names up to the first colon (:).
-+ size_t iColon = 0;
-+ do {
-+ port = (char *) ports[ nChannels ];
-+ iColon = port.find(":");
-+ if ( iColon != std::string::npos ) {
-+ port = port.substr( 0, iColon + 1 );
-+ if ( port != previousPort ) {
-+ nDevices++;
-+ previousPort = port;
-+ }
-+ }
-+ } while ( ports[++nChannels] );
-+ free( ports );
-+ }
-+
-+ jack_client_close( client );
-+ return nDevices;
-+}
-+
-+RtAudio::DeviceInfo RtApiJack :: getDeviceInfo( unsigned int device )
-+{
-+ RtAudio::DeviceInfo info;
-+ info.probed = false;
-+
-+ jack_options_t options = (jack_options_t) ( JackNoStartServer ); //JackNullOption
-+ jack_status_t *status = NULL;
-+ jack_client_t *client = jack_client_open( "RtApiJackInfo", options, status );
-+ if ( client == 0 ) {
-+ errorText_ = "RtApiJack::getDeviceInfo: Jack server not found or connection error!";
-+ error( RtAudioError::WARNING );
-+ return info;
-+ }
-+
-+ const char **ports;
-+ std::string port, previousPort;
-+ unsigned int nPorts = 0, nDevices = 0;
-+ ports = jack_get_ports( client, NULL, NULL, 0 );
-+ if ( ports ) {
-+ // Parse the port names up to the first colon (:).
-+ size_t iColon = 0;
-+ do {
-+ port = (char *) ports[ nPorts ];
-+ iColon = port.find(":");
-+ if ( iColon != std::string::npos ) {
-+ port = port.substr( 0, iColon );
-+ if ( port != previousPort ) {
-+ if ( nDevices == device ) info.name = port;
-+ nDevices++;
-+ previousPort = port;
-+ }
-+ }
-+ } while ( ports[++nPorts] );
-+ free( ports );
-+ }
-+
-+ if ( device >= nDevices ) {
-+ jack_client_close( client );
-+ errorText_ = "RtApiJack::getDeviceInfo: device ID is invalid!";
-+ error( RtAudioError::INVALID_USE );
-+ return info;
-+ }
-+
-+ // Get the current jack server sample rate.
-+ info.sampleRates.clear();
-+
-+ info.preferredSampleRate = jack_get_sample_rate( client );
-+ info.sampleRates.push_back( info.preferredSampleRate );
-+
-+ // Count the available ports containing the client name as device
-+ // channels. Jack "input ports" equal RtAudio output channels.
-+ unsigned int nChannels = 0;
-+ ports = jack_get_ports( client, info.name.c_str(), NULL, JackPortIsInput );
-+ if ( ports ) {
-+ while ( ports[ nChannels ] ) nChannels++;
-+ free( ports );
-+ info.outputChannels = nChannels;
-+ }
-+
-+ // Jack "output ports" equal RtAudio input channels.
-+ nChannels = 0;
-+ ports = jack_get_ports( client, info.name.c_str(), NULL, JackPortIsOutput );
-+ if ( ports ) {
-+ while ( ports[ nChannels ] ) nChannels++;
-+ free( ports );
-+ info.inputChannels = nChannels;
-+ }
-+
-+ if ( info.outputChannels == 0 && info.inputChannels == 0 ) {
-+ jack_client_close(client);
-+ errorText_ = "RtApiJack::getDeviceInfo: error determining Jack input/output channels!";
-+ error( RtAudioError::WARNING );
-+ return info;
-+ }
-+
-+ // If device opens for both playback and capture, we determine the channels.
-+ if ( info.outputChannels > 0 && info.inputChannels > 0 )
-+ info.duplexChannels = (info.outputChannels > info.inputChannels) ? info.inputChannels : info.outputChannels;
-+
-+ // Jack always uses 32-bit floats.
-+ info.nativeFormats = RTAUDIO_FLOAT32;
-+
-+ // Jack doesn't provide default devices so we'll use the first available one.
-+ if ( device == 0 && info.outputChannels > 0 )
-+ info.isDefaultOutput = true;
-+ if ( device == 0 && info.inputChannels > 0 )
-+ info.isDefaultInput = true;
-+
-+ jack_client_close(client);
-+ info.probed = true;
-+ return info;
-+}
-+
-+static int jackCallbackHandler( jack_nframes_t nframes, void *infoPointer )
-+{
-+ CallbackInfo *info = (CallbackInfo *) infoPointer;
-+
-+ RtApiJack *object = (RtApiJack *) info->object;
-+ if ( object->callbackEvent( (unsigned long) nframes ) == false ) return 1;
-+
-+ return 0;
-+}
-+
-+// This function will be called by a spawned thread when the Jack
-+// server signals that it is shutting down. It is necessary to handle
-+// it this way because the jackShutdown() function must return before
-+// the jack_deactivate() function (in closeStream()) will return.
-+static void *jackCloseStream( void *ptr )
-+{
-+ CallbackInfo *info = (CallbackInfo *) ptr;
-+ RtApiJack *object = (RtApiJack *) info->object;
-+
-+ object->closeStream();
-+
-+ pthread_exit( NULL );
-+}
-+static void jackShutdown( void *infoPointer )
-+{
-+ CallbackInfo *info = (CallbackInfo *) infoPointer;
-+ RtApiJack *object = (RtApiJack *) info->object;
-+
-+ // Check current stream state. If stopped, then we'll assume this
-+ // was called as a result of a call to RtApiJack::stopStream (the
-+ // deactivation of a client handle causes this function to be called).
-+ // If not, we'll assume the Jack server is shutting down or some
-+ // other problem occurred and we should close the stream.
-+ if ( object->isStreamRunning() == false ) return;
-+
-+ ThreadHandle threadId;
-+ pthread_create( &threadId, NULL, jackCloseStream, info );
-+ std::cerr << "\nRtApiJack: the Jack server is shutting down this client ... stream stopped and closed!!\n" << std::endl;
-+}
-+
-+static int jackXrun( void *infoPointer )
-+{
-+ JackHandle *handle = (JackHandle *) infoPointer;
-+
-+ if ( handle->ports[0] ) handle->xrun[0] = true;
-+ if ( handle->ports[1] ) handle->xrun[1] = true;
-+
-+ return 0;
-+}
-+
-+bool RtApiJack :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,
-+ unsigned int firstChannel, unsigned int sampleRate,
-+ RtAudioFormat format, unsigned int *bufferSize,
-+ RtAudio::StreamOptions *options )
-+{
-+ JackHandle *handle = (JackHandle *) stream_.apiHandle;
-+
-+ // Look for jack server and try to become a client (only do once per stream).
-+ jack_client_t *client = 0;
-+ if ( mode == OUTPUT || ( mode == INPUT && stream_.mode != OUTPUT ) ) {
-+ jack_options_t jackoptions = (jack_options_t) ( JackNoStartServer ); //JackNullOption;
-+ jack_status_t *status = NULL;
-+ if ( options && !options->streamName.empty() )
-+ client = jack_client_open( options->streamName.c_str(), jackoptions, status );
-+ else
-+ client = jack_client_open( "RtApiJack", jackoptions, status );
-+ if ( client == 0 ) {
-+ errorText_ = "RtApiJack::probeDeviceOpen: Jack server not found or connection error!";
-+ error( RtAudioError::WARNING );
-+ return FAILURE;
-+ }
-+ }
-+ else {
-+ // The handle must have been created on an earlier pass.
-+ client = handle->client;
-+ }
-+
-+ const char **ports;
-+ std::string port, previousPort, deviceName;
-+ unsigned int nPorts = 0, nDevices = 0;
-+ ports = jack_get_ports( client, NULL, NULL, 0 );
-+ if ( ports ) {
-+ // Parse the port names up to the first colon (:).
-+ size_t iColon = 0;
-+ do {
-+ port = (char *) ports[ nPorts ];
-+ iColon = port.find(":");
-+ if ( iColon != std::string::npos ) {
-+ port = port.substr( 0, iColon );
-+ if ( port != previousPort ) {
-+ if ( nDevices == device ) deviceName = port;
-+ nDevices++;
-+ previousPort = port;
-+ }
-+ }
-+ } while ( ports[++nPorts] );
-+ free( ports );
-+ }
-+
-+ if ( device >= nDevices ) {
-+ errorText_ = "RtApiJack::probeDeviceOpen: device ID is invalid!";
-+ return FAILURE;
-+ }
-+
-+ // Count the available ports containing the client name as device
-+ // channels. Jack "input ports" equal RtAudio output channels.
-+ unsigned int nChannels = 0;
-+ unsigned long flag = JackPortIsInput;
-+ if ( mode == INPUT ) flag = JackPortIsOutput;
-+ ports = jack_get_ports( client, deviceName.c_str(), NULL, flag );
-+ if ( ports ) {
-+ while ( ports[ nChannels ] ) nChannels++;
-+ free( ports );
-+ }
-+
-+ // Compare the jack ports for specified client to the requested number of channels.
-+ if ( nChannels < (channels + firstChannel) ) {
-+ errorStream_ << "RtApiJack::probeDeviceOpen: requested number of channels (" << channels << ") + offset (" << firstChannel << ") not found for specified device (" << device << ":" << deviceName << ").";
-+ errorText_ = errorStream_.str();
-+ return FAILURE;
-+ }
-+
-+ // Check the jack server sample rate.
-+ unsigned int jackRate = jack_get_sample_rate( client );
-+ if ( sampleRate != jackRate ) {
-+ jack_client_close( client );
-+ errorStream_ << "RtApiJack::probeDeviceOpen: the requested sample rate (" << sampleRate << ") is different than the JACK server rate (" << jackRate << ").";
-+ errorText_ = errorStream_.str();
-+ return FAILURE;
-+ }
-+ stream_.sampleRate = jackRate;
-+
-+ // Get the latency of the JACK port.
-+ ports = jack_get_ports( client, deviceName.c_str(), NULL, flag );
-+ if ( ports[ firstChannel ] ) {
-+ // Added by Ge Wang
-+ jack_latency_callback_mode_t cbmode = (mode == INPUT ? JackCaptureLatency : JackPlaybackLatency);
-+ // the range (usually the min and max are equal)
-+ jack_latency_range_t latrange; latrange.min = latrange.max = 0;
-+ // get the latency range
-+ jack_port_get_latency_range( jack_port_by_name( client, ports[firstChannel] ), cbmode, &latrange );
-+ // be optimistic, use the min!
-+ stream_.latency[mode] = latrange.min;
-+ //stream_.latency[mode] = jack_port_get_latency( jack_port_by_name( client, ports[ firstChannel ] ) );
-+ }
-+ free( ports );
-+
-+ // The jack server always uses 32-bit floating-point data.
-+ stream_.deviceFormat[mode] = RTAUDIO_FLOAT32;
-+ stream_.userFormat = format;
-+
-+ if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) stream_.userInterleaved = false;
-+ else stream_.userInterleaved = true;
-+
-+ // Jack always uses non-interleaved buffers.
-+ stream_.deviceInterleaved[mode] = false;
-+
-+ // Jack always provides host byte-ordered data.
-+ stream_.doByteSwap[mode] = false;
-+
-+ // Get the buffer size. The buffer size and number of buffers
-+ // (periods) is set when the jack server is started.
-+ stream_.bufferSize = (int) jack_get_buffer_size( client );
-+ *bufferSize = stream_.bufferSize;
-+
-+ stream_.nDeviceChannels[mode] = channels;
-+ stream_.nUserChannels[mode] = channels;
-+
-+ // Set flags for buffer conversion.
-+ stream_.doConvertBuffer[mode] = false;
-+ if ( stream_.userFormat != stream_.deviceFormat[mode] )
-+ stream_.doConvertBuffer[mode] = true;
-+ if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] &&
-+ stream_.nUserChannels[mode] > 1 )
-+ stream_.doConvertBuffer[mode] = true;
-+
-+ // Allocate our JackHandle structure for the stream.
-+ if ( handle == 0 ) {
-+ try {
-+ handle = new JackHandle;
-+ }
-+ catch ( std::bad_alloc& ) {
-+ errorText_ = "RtApiJack::probeDeviceOpen: error allocating JackHandle memory.";
-+ goto error;
-+ }
-+
-+ if ( pthread_cond_init(&handle->condition, NULL) ) {
-+ errorText_ = "RtApiJack::probeDeviceOpen: error initializing pthread condition variable.";
-+ goto error;
-+ }
-+ stream_.apiHandle = (void *) handle;
-+ handle->client = client;
-+ }
-+ handle->deviceName[mode] = deviceName;
-+
-+ // Allocate necessary internal buffers.
-+ unsigned long bufferBytes;
-+ bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat );
-+ stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 );
-+ if ( stream_.userBuffer[mode] == NULL ) {
-+ errorText_ = "RtApiJack::probeDeviceOpen: error allocating user buffer memory.";
-+ goto error;
-+ }
-+
-+ if ( stream_.doConvertBuffer[mode] ) {
-+
-+ bool makeBuffer = true;
-+ if ( mode == OUTPUT )
-+ bufferBytes = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] );
-+ else { // mode == INPUT
-+ bufferBytes = stream_.nDeviceChannels[1] * formatBytes( stream_.deviceFormat[1] );
-+ if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) {
-+ unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes(stream_.deviceFormat[0]);
-+ if ( bufferBytes < bytesOut ) makeBuffer = false;
-+ }
-+ }
-+
-+ if ( makeBuffer ) {
-+ bufferBytes *= *bufferSize;
-+ if ( stream_.deviceBuffer ) free( stream_.deviceBuffer );
-+ stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 );
-+ if ( stream_.deviceBuffer == NULL ) {
-+ errorText_ = "RtApiJack::probeDeviceOpen: error allocating device buffer memory.";
-+ goto error;
-+ }
-+ }
-+ }
-+
-+ // Allocate memory for the Jack ports (channels) identifiers.
-+ handle->ports[mode] = (jack_port_t **) malloc ( sizeof (jack_port_t *) * channels );
-+ if ( handle->ports[mode] == NULL ) {
-+ errorText_ = "RtApiJack::probeDeviceOpen: error allocating port memory.";
-+ goto error;
-+ }
-+
-+ stream_.device[mode] = device;
-+ stream_.channelOffset[mode] = firstChannel;
-+ stream_.state = STREAM_STOPPED;
-+ stream_.callbackInfo.object = (void *) this;
-+
-+ if ( stream_.mode == OUTPUT && mode == INPUT )
-+ // We had already set up the stream for output.
-+ stream_.mode = DUPLEX;
-+ else {
-+ stream_.mode = mode;
-+ jack_set_process_callback( handle->client, jackCallbackHandler, (void *) &stream_.callbackInfo );
-+ jack_set_xrun_callback( handle->client, jackXrun, (void *) &handle );
-+ jack_on_shutdown( handle->client, jackShutdown, (void *) &stream_.callbackInfo );
-+ }
-+
-+ // Register our ports.
-+ char label[64];
-+ if ( mode == OUTPUT ) {
-+ for ( unsigned int i=0; i<stream_.nUserChannels[0]; i++ ) {
-+ snprintf( label, 64, "outport %d", i );
-+ handle->ports[0][i] = jack_port_register( handle->client, (const char *)label,
-+ JACK_DEFAULT_AUDIO_TYPE, JackPortIsOutput, 0 );
-+ }
-+ }
-+ else {
-+ for ( unsigned int i=0; i<stream_.nUserChannels[1]; i++ ) {
-+ snprintf( label, 64, "inport %d", i );
-+ handle->ports[1][i] = jack_port_register( handle->client, (const char *)label,
-+ JACK_DEFAULT_AUDIO_TYPE, JackPortIsInput, 0 );
-+ }
-+ }
-+
-+ // Setup the buffer conversion information structure. We don't use
-+ // buffers to do channel offsets, so we override that parameter
-+ // here.
-+ if ( stream_.doConvertBuffer[mode] ) setConvertInfo( mode, 0 );
-+
-+ 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(), NULL, JackPortIsInput);
-+ if ( ports == NULL) {
-+ errorText_ = "RtApiJack::startStream(): error determining available JACK input ports!";
-+ goto unlock;
-+ }
-+
-+ // Now make the port connections. Since RtAudio wasn't designed to
-+ // allow the user to select particular channels of a device, we'll
-+ // just open the first "nChannels" ports with offset.
-+ for ( unsigned int i=0; i<stream_.nUserChannels[0]; i++ ) {
-+ result = 1;
-+ if ( ports[ stream_.channelOffset[0] + i ] )
-+ result = jack_connect( handle->client, jack_port_name( handle->ports[0][i] ), ports[ stream_.channelOffset[0] + i ] );
-+ if ( result ) {
-+ free( ports );
-+ errorText_ = "RtApiJack::startStream(): error connecting output ports!";
-+ goto unlock;
-+ }
-+ }
-+ free(ports);
-+ }
-+
-+ if ( shouldAutoconnect_ && (stream_.mode == INPUT || stream_.mode == DUPLEX) ) {
-+ result = 1;
-+ ports = jack_get_ports( handle->client, handle->deviceName[1].c_str(), NULL, JackPortIsOutput );
-+ if ( ports == NULL) {
-+ errorText_ = "RtApiJack::startStream(): error determining available JACK output ports!";
-+ goto unlock;
-+ }
-+
-+ // Now make the port connections. See note above.
-+ for ( unsigned int i=0; i<stream_.nUserChannels[1]; i++ ) {
-+ result = 1;
-+ if ( ports[ stream_.channelOffset[1] + i ] )
-+ result = jack_connect( handle->client, ports[ stream_.channelOffset[1] + i ], jack_port_name( handle->ports[1][i] ) );
-+ if ( result ) {
-+ free( ports );
-+ errorText_ = "RtApiJack::startStream(): error connecting input ports!";
-+ goto unlock;
-+ }
-+ }
-+ free(ports);
-+ }
-+
-+ handle->drainCounter = 0;
-+ handle->internalDrain = false;
-+ stream_.state = STREAM_RUNNING;
-+
-+ unlock:
-+ if ( result == 0 ) return;
-+ error( RtAudioError::SYSTEM_ERROR );
-+}
-+
-+void RtApiJack :: stopStream( void )
-+{
-+ verifyStream();
-+ if ( stream_.state == STREAM_STOPPED ) {
-+ errorText_ = "RtApiJack::stopStream(): the stream is already stopped!";
-+ error( RtAudioError::WARNING );
-+ return;
-+ }
-+
-+ JackHandle *handle = (JackHandle *) stream_.apiHandle;
-+ if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
-+
-+ if ( handle->drainCounter == 0 ) {
-+ handle->drainCounter = 2;
-+ pthread_cond_wait( &handle->condition, &stream_.mutex ); // block until signaled
-+ }
-+ }
-+
-+ jack_deactivate( handle->client );
-+ stream_.state = STREAM_STOPPED;
-+}
-+
-+void RtApiJack :: abortStream( void )
-+{
-+ verifyStream();
-+ if ( stream_.state == STREAM_STOPPED ) {
-+ errorText_ = "RtApiJack::abortStream(): the stream is already stopped!";
-+ error( RtAudioError::WARNING );
-+ return;
-+ }
-+
-+ JackHandle *handle = (JackHandle *) stream_.apiHandle;
-+ handle->drainCounter = 2;
-+
-+ stopStream();
-+}
-+
-+// This function will be called by a spawned thread when the user
-+// callback function signals that the stream should be stopped or
-+// aborted. It is necessary to handle it this way because the
-+// callbackEvent() function must return before the jack_deactivate()
-+// function will return.
-+static void *jackStopStream( void *ptr )
-+{
-+ CallbackInfo *info = (CallbackInfo *) ptr;
-+ RtApiJack *object = (RtApiJack *) info->object;
-+
-+ object->stopStream();
-+ pthread_exit( NULL );
-+}
-+
-+bool RtApiJack :: callbackEvent( unsigned long nframes )
-+{
-+ if ( stream_.state == STREAM_STOPPED || stream_.state == STREAM_STOPPING ) return SUCCESS;
-+ if ( stream_.state == STREAM_CLOSED ) {
-+ errorText_ = "RtApiCore::callbackEvent(): the stream is closed ... this shouldn't happen!";
-+ error( RtAudioError::WARNING );
-+ return FAILURE;
-+ }
-+ if ( stream_.bufferSize != nframes ) {
-+ errorText_ = "RtApiCore::callbackEvent(): the JACK buffer size has changed ... cannot process!";
-+ error( RtAudioError::WARNING );
-+ return FAILURE;
-+ }
-+
-+ CallbackInfo *info = (CallbackInfo *) &stream_.callbackInfo;
-+ JackHandle *handle = (JackHandle *) stream_.apiHandle;
-+
-+ // Check if we were draining the stream and signal is finished.
-+ if ( handle->drainCounter > 3 ) {
-+ ThreadHandle threadId;
-+
-+ stream_.state = STREAM_STOPPING;
-+ if ( handle->internalDrain == true )
-+ pthread_create( &threadId, NULL, jackStopStream, info );
-+ else
-+ pthread_cond_signal( &handle->condition );
-+ return SUCCESS;
-+ }
-+
-+ // Invoke user callback first, to get fresh output data.
-+ if ( handle->drainCounter == 0 ) {
-+ RtAudioCallback callback = (RtAudioCallback) info->callback;
-+ double streamTime = getStreamTime();
-+ RtAudioStreamStatus status = 0;
-+ if ( stream_.mode != INPUT && handle->xrun[0] == true ) {
-+ status |= RTAUDIO_OUTPUT_UNDERFLOW;
-+ handle->xrun[0] = false;
-+ }
-+ if ( stream_.mode != OUTPUT && handle->xrun[1] == true ) {
-+ status |= RTAUDIO_INPUT_OVERFLOW;
-+ handle->xrun[1] = false;
-+ }
-+ int cbReturnValue = callback( stream_.userBuffer[0], stream_.userBuffer[1],
-+ stream_.bufferSize, streamTime, status, info->userData );
-+ if ( cbReturnValue == 2 ) {
-+ stream_.state = STREAM_STOPPING;
-+ handle->drainCounter = 2;
-+ ThreadHandle id;
-+ pthread_create( &id, NULL, jackStopStream, info );
-+ return SUCCESS;
-+ }
-+ else if ( cbReturnValue == 1 ) {
-+ handle->drainCounter = 1;
-+ handle->internalDrain = true;
-+ }
-+ }
-+
-+ jack_default_audio_sample_t *jackbuffer;
-+ unsigned long bufferBytes = nframes * sizeof( jack_default_audio_sample_t );
-+ if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
-+
-+ if ( handle->drainCounter > 1 ) { // write zeros to the output stream
-+
-+ for ( unsigned int i=0; i<stream_.nDeviceChannels[0]; i++ ) {
-+ jackbuffer = (jack_default_audio_sample_t *) jack_port_get_buffer( handle->ports[0][i], (jack_nframes_t) nframes );
-+ memset( jackbuffer, 0, bufferBytes );
-+ }
-+
-+ }
-+ else if ( stream_.doConvertBuffer[0] ) {
-+
-+ convertBuffer( stream_.deviceBuffer, stream_.userBuffer[0], stream_.convertInfo[0] );
-+
-+ for ( unsigned int i=0; i<stream_.nDeviceChannels[0]; i++ ) {
-+ jackbuffer = (jack_default_audio_sample_t *) jack_port_get_buffer( handle->ports[0][i], (jack_nframes_t) nframes );
-+ memcpy( jackbuffer, &stream_.deviceBuffer[i*bufferBytes], bufferBytes );
-+ }
-+ }
-+ else { // no buffer conversion
-+ for ( unsigned int i=0; i<stream_.nUserChannels[0]; i++ ) {
-+ jackbuffer = (jack_default_audio_sample_t *) jack_port_get_buffer( handle->ports[0][i], (jack_nframes_t) nframes );
-+ memcpy( jackbuffer, &stream_.userBuffer[0][i*bufferBytes], bufferBytes );
-+ }
-+ }
-+ }
-+
-+ // Don't bother draining input
-+ if ( handle->drainCounter ) {
-+ handle->drainCounter++;
-+ goto unlock;
-+ }
-+
-+ if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {
-+
-+ if ( stream_.doConvertBuffer[1] ) {
-+ for ( unsigned int i=0; i<stream_.nDeviceChannels[1]; i++ ) {
-+ jackbuffer = (jack_default_audio_sample_t *) jack_port_get_buffer( handle->ports[1][i], (jack_nframes_t) nframes );
-+ memcpy( &stream_.deviceBuffer[i*bufferBytes], jackbuffer, bufferBytes );
-+ }
-+ convertBuffer( stream_.userBuffer[1], stream_.deviceBuffer, stream_.convertInfo[1] );
-+ }
-+ else { // no buffer conversion
-+ for ( unsigned int i=0; i<stream_.nUserChannels[1]; i++ ) {
-+ jackbuffer = (jack_default_audio_sample_t *) jack_port_get_buffer( handle->ports[1][i], (jack_nframes_t) nframes );
-+ memcpy( &stream_.userBuffer[1][i*bufferBytes], jackbuffer, bufferBytes );
-+ }
-+ }
-+ }
-+
-+ unlock:
-+ RtApi::tickStreamTime();
-+ return SUCCESS;
-+}
-+
-+ //******************** End of __UNIX_JACK__ *********************//
-+#endif
-+
-+#if defined(__WINDOWS_ASIO__) // ASIO API on Windows
-+
-+// The ASIO API is designed around a callback scheme, so this
-+// implementation is similar to that used for OS-X CoreAudio and Linux
-+// Jack. The primary constraint with ASIO is that it only allows
-+// access to a single driver at a time. Thus, it is not possible to
-+// have more than one simultaneous RtAudio stream.
-+//
-+// This implementation also requires a number of external ASIO files
-+// and a few global variables. The ASIO callback scheme does not
-+// allow for the passing of user data, so we must create a global
-+// pointer to our callbackInfo structure.
-+//
-+// On unix systems, we make use of a pthread condition variable.
-+// Since there is no equivalent in Windows, I hacked something based
-+// on information found in
-+// http://www.cs.wustl.edu/~schmidt/win32-cv-1.html.
-+
-+#include "asiosys.h"
-+#include "asio.h"
-+#include "iasiothiscallresolver.h"
-+#include "asiodrivers.h"
-+#include <cmath>
-+
-+static AsioDrivers drivers;
-+static ASIOCallbacks asioCallbacks;
-+static ASIODriverInfo driverInfo;
-+static CallbackInfo *asioCallbackInfo;
-+static bool asioXRun;
-+
-+struct AsioHandle {
-+ int drainCounter; // Tracks callback counts when draining
-+ bool internalDrain; // Indicates if stop is initiated from callback or not.
-+ ASIOBufferInfo *bufferInfos;
-+ HANDLE condition;
-+
-+ AsioHandle()
-+ :drainCounter(0), internalDrain(false), bufferInfos(0) {}
-+};
-+
-+// Function declarations (definitions at end of section)
-+static const char* getAsioErrorString( ASIOError result );
-+static void sampleRateChanged( ASIOSampleRate sRate );
-+static long asioMessages( long selector, long value, void* message, double* opt );
-+
-+RtApiAsio :: RtApiAsio()
-+{
-+ // ASIO cannot run on a multi-threaded appartment. You can call
-+ // CoInitialize beforehand, but it must be for appartment threading
-+ // (in which case, CoInitilialize will return S_FALSE here).
-+ coInitialized_ = false;
-+ HRESULT hr = CoInitialize( NULL );
-+ if ( FAILED(hr) ) {
-+ errorText_ = "RtApiAsio::ASIO requires a single-threaded appartment. Call CoInitializeEx(0,COINIT_APARTMENTTHREADED)";
-+ error( RtAudioError::WARNING );
-+ }
-+ coInitialized_ = true;
-+
-+ drivers.removeCurrentDriver();
-+ driverInfo.asioVersion = 2;
-+
-+ // See note in DirectSound implementation about GetDesktopWindow().
-+ driverInfo.sysRef = GetForegroundWindow();
-+}
-+
-+RtApiAsio :: ~RtApiAsio()
-+{
-+ if ( stream_.state != STREAM_CLOSED ) closeStream();
-+ if ( coInitialized_ ) CoUninitialize();
-+}
-+
-+unsigned int RtApiAsio :: getDeviceCount( void )
-+{
-+ return (unsigned int) drivers.asioGetNumDev();
-+}
-+
-+RtAudio::DeviceInfo RtApiAsio :: getDeviceInfo( unsigned int device )
-+{
-+ RtAudio::DeviceInfo info;
-+ info.probed = false;
-+
-+ // Get device ID
-+ unsigned int nDevices = getDeviceCount();
-+ if ( nDevices == 0 ) {
-+ errorText_ = "RtApiAsio::getDeviceInfo: no devices found!";
-+ error( RtAudioError::INVALID_USE );
-+ return info;
-+ }
-+
-+ if ( device >= nDevices ) {
-+ errorText_ = "RtApiAsio::getDeviceInfo: device ID is invalid!";
-+ error( RtAudioError::INVALID_USE );
-+ return info;
-+ }
-+
-+ // If a stream is already open, we cannot probe other devices. Thus, use the saved results.
-+ if ( stream_.state != STREAM_CLOSED ) {
-+ if ( device >= devices_.size() ) {
-+ errorText_ = "RtApiAsio::getDeviceInfo: device ID was not present before stream was opened.";
-+ error( RtAudioError::WARNING );
-+ return info;
-+ }
-+ return devices_[ device ];
-+ }
-+
-+ char driverName[32];
-+ ASIOError result = drivers.asioGetDriverName( (int) device, driverName, 32 );
-+ if ( result != ASE_OK ) {
-+ errorStream_ << "RtApiAsio::getDeviceInfo: unable to get driver name (" << getAsioErrorString( result ) << ").";
-+ errorText_ = errorStream_.str();
-+ error( RtAudioError::WARNING );
-+ return info;
-+ }
-+
-+ info.name = driverName;
-+
-+ if ( !drivers.loadDriver( driverName ) ) {
-+ errorStream_ << "RtApiAsio::getDeviceInfo: unable to load driver (" << driverName << ").";
-+ errorText_ = errorStream_.str();
-+ error( RtAudioError::WARNING );
-+ return info;
-+ }
-+
-+ result = ASIOInit( &driverInfo );
-+ if ( result != ASE_OK ) {
-+ errorStream_ << "RtApiAsio::getDeviceInfo: error (" << getAsioErrorString( result ) << ") initializing driver (" << driverName << ").";
-+ errorText_ = errorStream_.str();
-+ error( RtAudioError::WARNING );
-+ return info;
-+ }
-+
-+ // Determine the device channel information.
-+ long inputChannels, outputChannels;
-+ result = ASIOGetChannels( &inputChannels, &outputChannels );
-+ if ( result != ASE_OK ) {
-+ drivers.removeCurrentDriver();
-+ errorStream_ << "RtApiAsio::getDeviceInfo: error (" << getAsioErrorString( result ) << ") getting channel count (" << driverName << ").";
-+ errorText_ = errorStream_.str();
-+ error( RtAudioError::WARNING );
-+ return info;
-+ }
-+
-+ info.outputChannels = outputChannels;
-+ info.inputChannels = inputChannels;
-+ if ( info.outputChannels > 0 && info.inputChannels > 0 )
-+ info.duplexChannels = (info.outputChannels > info.inputChannels) ? info.inputChannels : info.outputChannels;
-+
-+ // Determine the supported sample rates.
-+ info.sampleRates.clear();
-+ for ( unsigned int i=0; i<MAX_SAMPLE_RATES; i++ ) {
-+ result = ASIOCanSampleRate( (ASIOSampleRate) SAMPLE_RATES[i] );
-+ if ( result == ASE_OK ) {
-+ info.sampleRates.push_back( SAMPLE_RATES[i] );
-+
-+ if ( !info.preferredSampleRate || ( SAMPLE_RATES[i] <= 48000 && SAMPLE_RATES[i] > info.preferredSampleRate ) )
-+ info.preferredSampleRate = SAMPLE_RATES[i];
-+ }
-+ }
-+
-+ // Determine supported data types ... just check first channel and assume rest are the same.
-+ ASIOChannelInfo channelInfo;
-+ channelInfo.channel = 0;
-+ channelInfo.isInput = true;
-+ if ( info.inputChannels <= 0 ) channelInfo.isInput = false;
-+ result = ASIOGetChannelInfo( &channelInfo );
-+ if ( result != ASE_OK ) {
-+ drivers.removeCurrentDriver();
-+ errorStream_ << "RtApiAsio::getDeviceInfo: error (" << getAsioErrorString( result ) << ") getting driver channel info (" << driverName << ").";
-+ errorText_ = errorStream_.str();
-+ error( RtAudioError::WARNING );
-+ return info;
-+ }
-+
-+ info.nativeFormats = 0;
-+ if ( channelInfo.type == ASIOSTInt16MSB || channelInfo.type == ASIOSTInt16LSB )
-+ info.nativeFormats |= RTAUDIO_SINT16;
-+ else if ( channelInfo.type == ASIOSTInt32MSB || channelInfo.type == ASIOSTInt32LSB )
-+ info.nativeFormats |= RTAUDIO_SINT32;
-+ else if ( channelInfo.type == ASIOSTFloat32MSB || channelInfo.type == ASIOSTFloat32LSB )
-+ info.nativeFormats |= RTAUDIO_FLOAT32;
-+ else if ( channelInfo.type == ASIOSTFloat64MSB || channelInfo.type == ASIOSTFloat64LSB )
-+ info.nativeFormats |= RTAUDIO_FLOAT64;
-+ else if ( channelInfo.type == ASIOSTInt24MSB || channelInfo.type == ASIOSTInt24LSB )
-+ info.nativeFormats |= RTAUDIO_SINT24;
-+
-+ if ( info.outputChannels > 0 )
-+ if ( getDefaultOutputDevice() == device ) info.isDefaultOutput = true;
-+ if ( info.inputChannels > 0 )
-+ if ( getDefaultInputDevice() == device ) info.isDefaultInput = true;
-+
-+ info.probed = true;
-+ drivers.removeCurrentDriver();
-+ return info;
-+}
-+
-+static void bufferSwitch( long index, ASIOBool /*processNow*/ )
-+{
-+ RtApiAsio *object = (RtApiAsio *) asioCallbackInfo->object;
-+ object->callbackEvent( index );
-+}
-+
-+void RtApiAsio :: saveDeviceInfo( void )
-+{
-+ devices_.clear();
-+
-+ unsigned int nDevices = getDeviceCount();
-+ devices_.resize( nDevices );
-+ for ( unsigned int i=0; i<nDevices; i++ )
-+ devices_[i] = getDeviceInfo( i );
-+}
-+
-+bool RtApiAsio :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,
-+ unsigned int firstChannel, unsigned int sampleRate,
-+ RtAudioFormat format, unsigned int *bufferSize,
-+ RtAudio::StreamOptions *options )
-+{////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////
-+
-+ bool isDuplexInput = mode == INPUT && stream_.mode == OUTPUT;
-+
-+ // For ASIO, a duplex stream MUST use the same driver.
-+ if ( isDuplexInput && stream_.device[0] != device ) {
-+ errorText_ = "RtApiAsio::probeDeviceOpen: an ASIO duplex stream must use the same device for input and output!";
-+ return FAILURE;
-+ }
-+
-+ char driverName[32];
-+ ASIOError result = drivers.asioGetDriverName( (int) device, driverName, 32 );
-+ if ( result != ASE_OK ) {
-+ errorStream_ << "RtApiAsio::probeDeviceOpen: unable to get driver name (" << getAsioErrorString( result ) << ").";
-+ errorText_ = errorStream_.str();
-+ return FAILURE;
-+ }
-+
-+ // Only load the driver once for duplex stream.
-+ if ( !isDuplexInput ) {
-+ // The getDeviceInfo() function will not work when a stream is open
-+ // because ASIO does not allow multiple devices to run at the same
-+ // time. Thus, we'll probe the system before opening a stream and
-+ // save the results for use by getDeviceInfo().
-+ this->saveDeviceInfo();
-+
-+ if ( !drivers.loadDriver( driverName ) ) {
-+ errorStream_ << "RtApiAsio::probeDeviceOpen: unable to load driver (" << driverName << ").";
-+ errorText_ = errorStream_.str();
-+ return FAILURE;
-+ }
-+
-+ result = ASIOInit( &driverInfo );
-+ if ( result != ASE_OK ) {
-+ errorStream_ << "RtApiAsio::probeDeviceOpen: error (" << getAsioErrorString( result ) << ") initializing driver (" << driverName << ").";
-+ errorText_ = errorStream_.str();
-+ return FAILURE;
-+ }
-+ }
-+
-+ // keep them before any "goto error", they are used for error cleanup + goto device boundary checks
-+ bool buffersAllocated = false;
-+ AsioHandle *handle = (AsioHandle *) stream_.apiHandle;
-+ unsigned int nChannels;
-+
-+
-+ // Check the device channel count.
-+ long inputChannels, outputChannels;
-+ result = ASIOGetChannels( &inputChannels, &outputChannels );
-+ if ( result != ASE_OK ) {
-+ errorStream_ << "RtApiAsio::probeDeviceOpen: error (" << getAsioErrorString( result ) << ") getting channel count (" << driverName << ").";
-+ errorText_ = errorStream_.str();
-+ goto error;
-+ }
-+
-+ if ( ( mode == OUTPUT && (channels+firstChannel) > (unsigned int) outputChannels) ||
-+ ( mode == INPUT && (channels+firstChannel) > (unsigned int) inputChannels) ) {
-+ errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") does not support requested channel count (" << channels << ") + offset (" << firstChannel << ").";
-+ errorText_ = errorStream_.str();
-+ goto error;
-+ }
-+ stream_.nDeviceChannels[mode] = channels;
-+ stream_.nUserChannels[mode] = channels;
-+ stream_.channelOffset[mode] = firstChannel;
-+
-+ // Verify the sample rate is supported.
-+ result = ASIOCanSampleRate( (ASIOSampleRate) sampleRate );
-+ if ( result != ASE_OK ) {
-+ errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") does not support requested sample rate (" << sampleRate << ").";
-+ errorText_ = errorStream_.str();
-+ goto error;
-+ }
-+
-+ // Get the current sample rate
-+ ASIOSampleRate currentRate;
-+ result = ASIOGetSampleRate( ¤tRate );
-+ if ( result != ASE_OK ) {
-+ errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error getting sample rate.";
-+ errorText_ = errorStream_.str();
-+ goto error;
-+ }
-+
-+ // Set the sample rate only if necessary
-+ if ( currentRate != sampleRate ) {
-+ result = ASIOSetSampleRate( (ASIOSampleRate) sampleRate );
-+ if ( result != ASE_OK ) {
-+ errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error setting sample rate (" << sampleRate << ").";
-+ errorText_ = errorStream_.str();
-+ goto error;
-+ }
-+ }
-+
-+ // Determine the driver data type.
-+ ASIOChannelInfo channelInfo;
-+ channelInfo.channel = 0;
-+ if ( mode == OUTPUT ) channelInfo.isInput = false;
-+ else channelInfo.isInput = true;
-+ result = ASIOGetChannelInfo( &channelInfo );
-+ if ( result != ASE_OK ) {
-+ errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error (" << getAsioErrorString( result ) << ") getting data format.";
-+ errorText_ = errorStream_.str();
-+ goto error;
-+ }
-+
-+ // Assuming WINDOWS host is always little-endian.
-+ stream_.doByteSwap[mode] = false;
-+ stream_.userFormat = format;
-+ stream_.deviceFormat[mode] = 0;
-+ if ( channelInfo.type == ASIOSTInt16MSB || channelInfo.type == ASIOSTInt16LSB ) {
-+ stream_.deviceFormat[mode] = RTAUDIO_SINT16;
-+ if ( channelInfo.type == ASIOSTInt16MSB ) stream_.doByteSwap[mode] = true;
-+ }
-+ else if ( channelInfo.type == ASIOSTInt32MSB || channelInfo.type == ASIOSTInt32LSB ) {
-+ stream_.deviceFormat[mode] = RTAUDIO_SINT32;
-+ if ( channelInfo.type == ASIOSTInt32MSB ) stream_.doByteSwap[mode] = true;
-+ }
-+ else if ( channelInfo.type == ASIOSTFloat32MSB || channelInfo.type == ASIOSTFloat32LSB ) {
-+ stream_.deviceFormat[mode] = RTAUDIO_FLOAT32;
-+ if ( channelInfo.type == ASIOSTFloat32MSB ) stream_.doByteSwap[mode] = true;
-+ }
-+ else if ( channelInfo.type == ASIOSTFloat64MSB || channelInfo.type == ASIOSTFloat64LSB ) {
-+ stream_.deviceFormat[mode] = RTAUDIO_FLOAT64;
-+ if ( channelInfo.type == ASIOSTFloat64MSB ) stream_.doByteSwap[mode] = true;
-+ }
-+ else if ( channelInfo.type == ASIOSTInt24MSB || channelInfo.type == ASIOSTInt24LSB ) {
-+ stream_.deviceFormat[mode] = RTAUDIO_SINT24;
-+ if ( channelInfo.type == ASIOSTInt24MSB ) stream_.doByteSwap[mode] = true;
-+ }
-+
-+ if ( stream_.deviceFormat[mode] == 0 ) {
-+ errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") data format not supported by RtAudio.";
-+ errorText_ = errorStream_.str();
-+ goto error;
-+ }
-+
-+ // Set the buffer size. For a duplex stream, this will end up
-+ // setting the buffer size based on the input constraints, which
-+ // should be ok.
-+ long minSize, maxSize, preferSize, granularity;
-+ result = ASIOGetBufferSize( &minSize, &maxSize, &preferSize, &granularity );
-+ if ( result != ASE_OK ) {
-+ errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error (" << getAsioErrorString( result ) << ") getting buffer size.";
-+ errorText_ = errorStream_.str();
-+ goto error;
-+ }
-+
-+ if ( isDuplexInput ) {
-+ // When this is the duplex input (output was opened before), then we have to use the same
-+ // buffersize as the output, because it might use the preferred buffer size, which most
-+ // likely wasn't passed as input to this. The buffer sizes have to be identically anyway,
-+ // So instead of throwing an error, make them equal. The caller uses the reference
-+ // to the "bufferSize" param as usual to set up processing buffers.
-+
-+ *bufferSize = stream_.bufferSize;
-+
-+ } else {
-+ if ( *bufferSize == 0 ) *bufferSize = preferSize;
-+ else if ( *bufferSize < (unsigned int) minSize ) *bufferSize = (unsigned int) minSize;
-+ else if ( *bufferSize > (unsigned int) maxSize ) *bufferSize = (unsigned int) maxSize;
-+ else if ( granularity == -1 ) {
-+ // Make sure bufferSize is a power of two.
-+ int log2_of_min_size = 0;
-+ int log2_of_max_size = 0;
-+
-+ for ( unsigned int i = 0; i < sizeof(long) * 8; i++ ) {
-+ if ( minSize & ((long)1 << i) ) log2_of_min_size = i;
-+ if ( maxSize & ((long)1 << i) ) log2_of_max_size = i;
-+ }
-+
-+ long min_delta = std::abs( (long)*bufferSize - ((long)1 << log2_of_min_size) );
-+ int min_delta_num = log2_of_min_size;
-+
-+ for (int i = log2_of_min_size + 1; i <= log2_of_max_size; i++) {
-+ long current_delta = std::abs( (long)*bufferSize - ((long)1 << i) );
-+ if (current_delta < min_delta) {
-+ min_delta = current_delta;
-+ min_delta_num = i;
-+ }
-+ }
-+
-+ *bufferSize = ( (unsigned int)1 << min_delta_num );
-+ if ( *bufferSize < (unsigned int) minSize ) *bufferSize = (unsigned int) minSize;
-+ else if ( *bufferSize > (unsigned int) maxSize ) *bufferSize = (unsigned int) maxSize;
-+ }
-+ else if ( granularity != 0 ) {
-+ // Set to an even multiple of granularity, rounding up.
-+ *bufferSize = (*bufferSize + granularity-1) / granularity * granularity;
-+ }
-+ }
-+
-+ /*
-+ // we don't use it anymore, see above!
-+ // Just left it here for the case...
-+ if ( isDuplexInput && stream_.bufferSize != *bufferSize ) {
-+ errorText_ = "RtApiAsio::probeDeviceOpen: input/output buffersize discrepancy!";
-+ goto error;
-+ }
-+ */
-+
-+ stream_.bufferSize = *bufferSize;
-+ stream_.nBuffers = 2;
-+
-+ if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) stream_.userInterleaved = false;
-+ else stream_.userInterleaved = true;
-+
-+ // ASIO always uses non-interleaved buffers.
-+ stream_.deviceInterleaved[mode] = false;
-+
-+ // Allocate, if necessary, our AsioHandle structure for the stream.
-+ if ( handle == 0 ) {
-+ try {
-+ handle = new AsioHandle;
-+ }
-+ catch ( std::bad_alloc& ) {
-+ errorText_ = "RtApiAsio::probeDeviceOpen: error allocating AsioHandle memory.";
-+ goto error;
-+ }
-+ handle->bufferInfos = 0;
-+
-+ // Create a manual-reset event.
-+ handle->condition = CreateEvent( NULL, // no security
-+ TRUE, // manual-reset
-+ FALSE, // non-signaled initially
-+ NULL ); // unnamed
-+ stream_.apiHandle = (void *) handle;
-+ }
-+
-+ // Create the ASIO internal buffers. Since RtAudio sets up input
-+ // and output separately, we'll have to dispose of previously
-+ // created output buffers for a duplex stream.
-+ if ( mode == INPUT && stream_.mode == OUTPUT ) {
-+ ASIODisposeBuffers();
-+ if ( handle->bufferInfos ) free( handle->bufferInfos );
-+ }
-+
-+ // Allocate, initialize, and save the bufferInfos in our stream callbackInfo structure.
-+ unsigned int i;
-+ nChannels = stream_.nDeviceChannels[0] + stream_.nDeviceChannels[1];
-+ handle->bufferInfos = (ASIOBufferInfo *) malloc( nChannels * sizeof(ASIOBufferInfo) );
-+ if ( handle->bufferInfos == NULL ) {
-+ errorStream_ << "RtApiAsio::probeDeviceOpen: error allocating bufferInfo memory for driver (" << driverName << ").";
-+ errorText_ = errorStream_.str();
-+ goto error;
-+ }
-+
-+ ASIOBufferInfo *infos;
-+ infos = handle->bufferInfos;
-+ for ( i=0; i<stream_.nDeviceChannels[0]; i++, infos++ ) {
-+ infos->isInput = ASIOFalse;
-+ infos->channelNum = i + stream_.channelOffset[0];
-+ infos->buffers[0] = infos->buffers[1] = 0;
-+ }
-+ for ( i=0; i<stream_.nDeviceChannels[1]; i++, infos++ ) {
-+ infos->isInput = ASIOTrue;
-+ infos->channelNum = i + stream_.channelOffset[1];
-+ infos->buffers[0] = infos->buffers[1] = 0;
-+ }
-+
-+ // prepare for callbacks
-+ stream_.sampleRate = sampleRate;
-+ stream_.device[mode] = device;
-+ stream_.mode = isDuplexInput ? DUPLEX : mode;
-+
-+ // store this class instance before registering callbacks, that are going to use it
-+ asioCallbackInfo = &stream_.callbackInfo;
-+ stream_.callbackInfo.object = (void *) this;
-+
-+ // Set up the ASIO callback structure and create the ASIO data buffers.
-+ asioCallbacks.bufferSwitch = &bufferSwitch;
-+ asioCallbacks.sampleRateDidChange = &sampleRateChanged;
-+ asioCallbacks.asioMessage = &asioMessages;
-+ asioCallbacks.bufferSwitchTimeInfo = NULL;
-+ result = ASIOCreateBuffers( handle->bufferInfos, nChannels, stream_.bufferSize, &asioCallbacks );
-+ if ( result != ASE_OK ) {
-+ // Standard method failed. This can happen with strict/misbehaving drivers that return valid buffer size ranges
-+ // but only accept the preferred buffer size as parameter for ASIOCreateBuffers. eg. Creatives ASIO driver
-+ // in that case, let's be naïve and try that instead
-+ *bufferSize = preferSize;
-+ stream_.bufferSize = *bufferSize;
-+ result = ASIOCreateBuffers( handle->bufferInfos, nChannels, stream_.bufferSize, &asioCallbacks );
-+ }
-+
-+ if ( result != ASE_OK ) {
-+ errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error (" << getAsioErrorString( result ) << ") creating buffers.";
-+ errorText_ = errorStream_.str();
-+ goto error;
-+ }
-+ buffersAllocated = true;
-+ stream_.state = STREAM_STOPPED;
-+
-+ // Set flags for buffer conversion.
-+ stream_.doConvertBuffer[mode] = false;
-+ if ( stream_.userFormat != stream_.deviceFormat[mode] )
-+ stream_.doConvertBuffer[mode] = true;
-+ if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] &&
-+ stream_.nUserChannels[mode] > 1 )
-+ stream_.doConvertBuffer[mode] = true;
-+
-+ // Allocate necessary internal buffers
-+ unsigned long bufferBytes;
-+ bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat );
-+ stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 );
-+ if ( stream_.userBuffer[mode] == NULL ) {
-+ errorText_ = "RtApiAsio::probeDeviceOpen: error allocating user buffer memory.";
-+ goto error;
-+ }
-+
-+ if ( stream_.doConvertBuffer[mode] ) {
-+
-+ bool makeBuffer = true;
-+ bufferBytes = stream_.nDeviceChannels[mode] * formatBytes( stream_.deviceFormat[mode] );
-+ if ( isDuplexInput && stream_.deviceBuffer ) {
-+ unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] );
-+ if ( bufferBytes <= bytesOut ) makeBuffer = false;
-+ }
-+
-+ if ( makeBuffer ) {
-+ bufferBytes *= *bufferSize;
-+ if ( stream_.deviceBuffer ) free( stream_.deviceBuffer );
-+ stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 );
-+ if ( stream_.deviceBuffer == NULL ) {
-+ errorText_ = "RtApiAsio::probeDeviceOpen: error allocating device buffer memory.";
-+ goto error;
-+ }
-+ }
-+ }
-+
-+ // Determine device latencies
-+ long inputLatency, outputLatency;
-+ result = ASIOGetLatencies( &inputLatency, &outputLatency );
-+ if ( result != ASE_OK ) {
-+ errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error (" << getAsioErrorString( result ) << ") getting latency.";
-+ errorText_ = errorStream_.str();
-+ error( RtAudioError::WARNING); // warn but don't fail
-+ }
-+ else {
-+ stream_.latency[0] = outputLatency;
-+ stream_.latency[1] = inputLatency;
-+ }
-+
-+ // Setup the buffer conversion information structure. We don't use
-+ // buffers to do channel offsets, so we override that parameter
-+ // here.
-+ if ( stream_.doConvertBuffer[mode] ) setConvertInfo( mode, 0 );
-+
-+ return SUCCESS;
-+
-+ error:
-+ if ( !isDuplexInput ) {
-+ // the cleanup for error in the duplex input, is done by RtApi::openStream
-+ // So we clean up for single channel only
-+
-+ if ( buffersAllocated )
-+ ASIODisposeBuffers();
-+
-+ drivers.removeCurrentDriver();
-+
-+ if ( handle ) {
-+ CloseHandle( handle->condition );
-+ if ( handle->bufferInfos )
-+ free( handle->bufferInfos );
-+
-+ delete handle;
-+ stream_.apiHandle = 0;
-+ }
-+
-+
-+ if ( stream_.userBuffer[mode] ) {
-+ free( stream_.userBuffer[mode] );
-+ stream_.userBuffer[mode] = 0;
-+ }
-+
-+ if ( stream_.deviceBuffer ) {
-+ free( stream_.deviceBuffer );
-+ stream_.deviceBuffer = 0;
-+ }
-+ }
-+
-+ return FAILURE;
-+}////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////
-+
-+void RtApiAsio :: closeStream()
-+{
-+ if ( stream_.state == STREAM_CLOSED ) {
-+ errorText_ = "RtApiAsio::closeStream(): no open stream to close!";
-+ error( RtAudioError::WARNING );
-+ return;
-+ }
-+
-+ if ( stream_.state == STREAM_RUNNING ) {
-+ stream_.state = STREAM_STOPPED;
-+ ASIOStop();
-+ }
-+ ASIODisposeBuffers();
-+ drivers.removeCurrentDriver();
-+
-+ AsioHandle *handle = (AsioHandle *) stream_.apiHandle;
-+ if ( handle ) {
-+ CloseHandle( handle->condition );
-+ if ( handle->bufferInfos )
-+ free( handle->bufferInfos );
-+ delete handle;
-+ stream_.apiHandle = 0;
-+ }
-+
-+ for ( int i=0; i<2; i++ ) {
-+ if ( stream_.userBuffer[i] ) {
-+ free( stream_.userBuffer[i] );
-+ stream_.userBuffer[i] = 0;
-+ }
-+ }
-+
-+ if ( stream_.deviceBuffer ) {
-+ free( stream_.deviceBuffer );
-+ stream_.deviceBuffer = 0;
-+ }
-+
-+ stream_.mode = UNINITIALIZED;
-+ stream_.state = STREAM_CLOSED;
-+}
-+
-+bool stopThreadCalled = false;
-+
-+void RtApiAsio :: startStream()
-+{
-+ verifyStream();
-+ if ( stream_.state == STREAM_RUNNING ) {
-+ errorText_ = "RtApiAsio::startStream(): the stream is already running!";
-+ error( RtAudioError::WARNING );
-+ return;
-+ }
-+
-+ AsioHandle *handle = (AsioHandle *) stream_.apiHandle;
-+ ASIOError result = ASIOStart();
-+ if ( result != ASE_OK ) {
-+ errorStream_ << "RtApiAsio::startStream: error (" << getAsioErrorString( result ) << ") starting device.";
-+ errorText_ = errorStream_.str();
-+ goto unlock;
-+ }
-+
-+ handle->drainCounter = 0;
-+ handle->internalDrain = false;
-+ ResetEvent( handle->condition );
-+ stream_.state = STREAM_RUNNING;
-+ asioXRun = false;
-+
-+ unlock:
-+ stopThreadCalled = false;
-+
-+ if ( result == ASE_OK ) return;
-+ error( RtAudioError::SYSTEM_ERROR );
-+}
-+
-+void RtApiAsio :: stopStream()
-+{
-+ verifyStream();
-+ if ( stream_.state == STREAM_STOPPED ) {
-+ errorText_ = "RtApiAsio::stopStream(): the stream is already stopped!";
-+ error( RtAudioError::WARNING );
-+ return;
-+ }
-+
-+ AsioHandle *handle = (AsioHandle *) stream_.apiHandle;
-+ if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
-+ if ( handle->drainCounter == 0 ) {
-+ handle->drainCounter = 2;
-+ WaitForSingleObject( handle->condition, INFINITE ); // block until signaled
-+ }
-+ }
-+
-+ stream_.state = STREAM_STOPPED;
-+
-+ ASIOError result = ASIOStop();
-+ if ( result != ASE_OK ) {
-+ errorStream_ << "RtApiAsio::stopStream: error (" << getAsioErrorString( result ) << ") stopping device.";
-+ errorText_ = errorStream_.str();
-+ }
-+
-+ if ( result == ASE_OK ) return;
-+ error( RtAudioError::SYSTEM_ERROR );
-+}
-+
-+void RtApiAsio :: abortStream()
-+{
-+ verifyStream();
-+ if ( stream_.state == STREAM_STOPPED ) {
-+ errorText_ = "RtApiAsio::abortStream(): the stream is already stopped!";
-+ error( RtAudioError::WARNING );
-+ return;
-+ }
-+
-+ // The following lines were commented-out because some behavior was
-+ // noted where the device buffers need to be zeroed to avoid
-+ // continuing sound, even when the device buffers are completely
-+ // disposed. So now, calling abort is the same as calling stop.
-+ // AsioHandle *handle = (AsioHandle *) stream_.apiHandle;
-+ // handle->drainCounter = 2;
-+ stopStream();
-+}
-+
-+// This function will be called by a spawned thread when the user
-+// callback function signals that the stream should be stopped or
-+// aborted. It is necessary to handle it this way because the
-+// callbackEvent() function must return before the ASIOStop()
-+// function will return.
-+static unsigned __stdcall asioStopStream( void *ptr )
-+{
-+ CallbackInfo *info = (CallbackInfo *) ptr;
-+ RtApiAsio *object = (RtApiAsio *) info->object;
-+
-+ object->stopStream();
-+ _endthreadex( 0 );
-+ return 0;
-+}
-+
-+bool RtApiAsio :: callbackEvent( long bufferIndex )
-+{
-+ if ( stream_.state == STREAM_STOPPED || stream_.state == STREAM_STOPPING ) return SUCCESS;
-+ if ( stream_.state == STREAM_CLOSED ) {
-+ errorText_ = "RtApiAsio::callbackEvent(): the stream is closed ... this shouldn't happen!";
-+ error( RtAudioError::WARNING );
-+ return FAILURE;
-+ }
-+
-+ CallbackInfo *info = (CallbackInfo *) &stream_.callbackInfo;
-+ AsioHandle *handle = (AsioHandle *) stream_.apiHandle;
-+
-+ // Check if we were draining the stream and signal if finished.
-+ if ( handle->drainCounter > 3 ) {
-+
-+ stream_.state = STREAM_STOPPING;
-+ if ( handle->internalDrain == false )
-+ SetEvent( handle->condition );
-+ else { // spawn a thread to stop the stream
-+ unsigned threadId;
-+ stream_.callbackInfo.thread = _beginthreadex( NULL, 0, &asioStopStream,
-+ &stream_.callbackInfo, 0, &threadId );
-+ }
-+ return SUCCESS;
-+ }
-+
-+ // Invoke user callback to get fresh output data UNLESS we are
-+ // draining stream.
-+ if ( handle->drainCounter == 0 ) {
-+ RtAudioCallback callback = (RtAudioCallback) info->callback;
-+ double streamTime = getStreamTime();
-+ RtAudioStreamStatus status = 0;
-+ if ( stream_.mode != INPUT && asioXRun == true ) {
-+ status |= RTAUDIO_OUTPUT_UNDERFLOW;
-+ asioXRun = false;
-+ }
-+ if ( stream_.mode != OUTPUT && asioXRun == true ) {
-+ status |= RTAUDIO_INPUT_OVERFLOW;
-+ asioXRun = false;
-+ }
-+ int cbReturnValue = callback( stream_.userBuffer[0], stream_.userBuffer[1],
-+ stream_.bufferSize, streamTime, status, info->userData );
-+ if ( cbReturnValue == 2 ) {
-+ stream_.state = STREAM_STOPPING;
-+ handle->drainCounter = 2;
-+ unsigned threadId;
-+ stream_.callbackInfo.thread = _beginthreadex( NULL, 0, &asioStopStream,
-+ &stream_.callbackInfo, 0, &threadId );
-+ return SUCCESS;
-+ }
-+ else if ( cbReturnValue == 1 ) {
-+ handle->drainCounter = 1;
-+ handle->internalDrain = true;
-+ }
-+ }
-+
-+ unsigned int nChannels, bufferBytes, i, j;
-+ nChannels = stream_.nDeviceChannels[0] + stream_.nDeviceChannels[1];
-+ if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
-+
-+ bufferBytes = stream_.bufferSize * formatBytes( stream_.deviceFormat[0] );
-+
-+ if ( handle->drainCounter > 1 ) { // write zeros to the output stream
-+
-+ for ( i=0, j=0; i<nChannels; i++ ) {
-+ if ( handle->bufferInfos[i].isInput != ASIOTrue )
-+ memset( handle->bufferInfos[i].buffers[bufferIndex], 0, bufferBytes );
-+ }
-+
-+ }
-+ else if ( stream_.doConvertBuffer[0] ) {
-+
-+ convertBuffer( stream_.deviceBuffer, stream_.userBuffer[0], stream_.convertInfo[0] );
-+ if ( stream_.doByteSwap[0] )
-+ byteSwapBuffer( stream_.deviceBuffer,
-+ stream_.bufferSize * stream_.nDeviceChannels[0],
-+ stream_.deviceFormat[0] );
-+
-+ for ( i=0, j=0; i<nChannels; i++ ) {
-+ if ( handle->bufferInfos[i].isInput != ASIOTrue )
-+ memcpy( handle->bufferInfos[i].buffers[bufferIndex],
-+ &stream_.deviceBuffer[j++*bufferBytes], bufferBytes );
-+ }
-+
-+ }
-+ else {
-+
-+ if ( stream_.doByteSwap[0] )
-+ byteSwapBuffer( stream_.userBuffer[0],
-+ stream_.bufferSize * stream_.nUserChannels[0],
-+ stream_.userFormat );
-+
-+ for ( i=0, j=0; i<nChannels; i++ ) {
-+ if ( handle->bufferInfos[i].isInput != ASIOTrue )
-+ memcpy( handle->bufferInfos[i].buffers[bufferIndex],
-+ &stream_.userBuffer[0][bufferBytes*j++], bufferBytes );
-+ }
-+
-+ }
-+ }
-+
-+ // Don't bother draining input
-+ if ( handle->drainCounter ) {
-+ handle->drainCounter++;
-+ goto unlock;
-+ }
-+
-+ if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {
-+
-+ bufferBytes = stream_.bufferSize * formatBytes(stream_.deviceFormat[1]);
-+
-+ if (stream_.doConvertBuffer[1]) {
-+
-+ // Always interleave ASIO input data.
-+ for ( i=0, j=0; i<nChannels; i++ ) {
-+ if ( handle->bufferInfos[i].isInput == ASIOTrue )
-+ memcpy( &stream_.deviceBuffer[j++*bufferBytes],
-+ handle->bufferInfos[i].buffers[bufferIndex],
-+ bufferBytes );
-+ }
-+
-+ if ( stream_.doByteSwap[1] )
-+ byteSwapBuffer( stream_.deviceBuffer,
-+ stream_.bufferSize * stream_.nDeviceChannels[1],
-+ stream_.deviceFormat[1] );
-+ convertBuffer( stream_.userBuffer[1], stream_.deviceBuffer, stream_.convertInfo[1] );
-+
-+ }
-+ else {
-+ for ( i=0, j=0; i<nChannels; i++ ) {
-+ if ( handle->bufferInfos[i].isInput == ASIOTrue ) {
-+ memcpy( &stream_.userBuffer[1][bufferBytes*j++],
-+ handle->bufferInfos[i].buffers[bufferIndex],
-+ bufferBytes );
-+ }
-+ }
-+
-+ if ( stream_.doByteSwap[1] )
-+ byteSwapBuffer( stream_.userBuffer[1],
-+ stream_.bufferSize * stream_.nUserChannels[1],
-+ stream_.userFormat );
-+ }
-+ }
-+
-+ unlock:
-+ // The following call was suggested by Malte Clasen. While the API
-+ // documentation indicates it should not be required, some device
-+ // drivers apparently do not function correctly without it.
-+ ASIOOutputReady();
-+
-+ RtApi::tickStreamTime();
-+ return SUCCESS;
-+}
-+
-+static void sampleRateChanged( ASIOSampleRate sRate )
-+{
-+ // The ASIO documentation says that this usually only happens during
-+ // external sync. Audio processing is not stopped by the driver,
-+ // actual sample rate might not have even changed, maybe only the
-+ // sample rate status of an AES/EBU or S/PDIF digital input at the
-+ // audio device.
-+
-+ RtApi *object = (RtApi *) asioCallbackInfo->object;
-+ try {
-+ object->stopStream();
-+ }
-+ catch ( RtAudioError &exception ) {
-+ std::cerr << "\nRtApiAsio: sampleRateChanged() error (" << exception.getMessage() << ")!\n" << std::endl;
-+ return;
-+ }
-+
-+ std::cerr << "\nRtApiAsio: driver reports sample rate changed to " << sRate << " ... stream stopped!!!\n" << std::endl;
-+}
-+
-+static long asioMessages( long selector, long value, void* /*message*/, double* /*opt*/ )
-+{
-+ long ret = 0;
-+
-+ switch( selector ) {
-+ case kAsioSelectorSupported:
-+ if ( value == kAsioResetRequest
-+ || value == kAsioEngineVersion
-+ || value == kAsioResyncRequest
-+ || value == kAsioLatenciesChanged
-+ // The following three were added for ASIO 2.0, you don't
-+ // necessarily have to support them.
-+ || value == kAsioSupportsTimeInfo
-+ || value == kAsioSupportsTimeCode
-+ || value == kAsioSupportsInputMonitor)
-+ ret = 1L;
-+ break;
-+ case kAsioResetRequest:
-+ // Defer the task and perform the reset of the driver during the
-+ // next "safe" situation. You cannot reset the driver right now,
-+ // as this code is called from the driver. Reset the driver is
-+ // done by completely destruct is. I.e. ASIOStop(),
-+ // ASIODisposeBuffers(), Destruction Afterwards you initialize the
-+ // driver again.
-+ std::cerr << "\nRtApiAsio: driver reset requested!!!" << std::endl;
-+ ret = 1L;
-+ break;
-+ case kAsioResyncRequest:
-+ // This informs the application that the driver encountered some
-+ // non-fatal data loss. It is used for synchronization purposes
-+ // of different media. Added mainly to work around the Win16Mutex
-+ // problems in Windows 95/98 with the Windows Multimedia system,
-+ // which could lose data because the Mutex was held too long by
-+ // another thread. However a driver can issue it in other
-+ // situations, too.
-+ // std::cerr << "\nRtApiAsio: driver resync requested!!!" << std::endl;
-+ asioXRun = true;
-+ ret = 1L;
-+ break;
-+ case kAsioLatenciesChanged:
-+ // This will inform the host application that the drivers were
-+ // latencies changed. Beware, it this does not mean that the
-+ // buffer sizes have changed! You might need to update internal
-+ // delay data.
-+ std::cerr << "\nRtApiAsio: driver latency may have changed!!!" << std::endl;
-+ ret = 1L;
-+ break;
-+ case kAsioEngineVersion:
-+ // Return the supported ASIO version of the host application. If
-+ // a host application does not implement this selector, ASIO 1.0
-+ // is assumed by the driver.
-+ ret = 2L;
-+ break;
-+ case kAsioSupportsTimeInfo:
-+ // Informs the driver whether the
-+ // asioCallbacks.bufferSwitchTimeInfo() callback is supported.
-+ // For compatibility with ASIO 1.0 drivers the host application
-+ // should always support the "old" bufferSwitch method, too.
-+ ret = 0;
-+ break;
-+ case kAsioSupportsTimeCode:
-+ // Informs the driver whether application is interested in time
-+ // code info. If an application does not need to know about time
-+ // code, the driver has less work to do.
-+ ret = 0;
-+ break;
-+ }
-+ return ret;
-+}
-+
-+static const char* getAsioErrorString( ASIOError result )
-+{
-+ struct Messages
-+ {
-+ ASIOError value;
-+ const char*message;
-+ };
-+
-+ static const Messages m[] =
-+ {
-+ { ASE_NotPresent, "Hardware input or output is not present or available." },
-+ { ASE_HWMalfunction, "Hardware is malfunctioning." },
-+ { ASE_InvalidParameter, "Invalid input parameter." },
-+ { ASE_InvalidMode, "Invalid mode." },
-+ { ASE_SPNotAdvancing, "Sample position not advancing." },
-+ { ASE_NoClock, "Sample clock or rate cannot be determined or is not present." },
-+ { ASE_NoMemory, "Not enough memory to complete the request." }
-+ };
-+
-+ for ( unsigned int i = 0; i < sizeof(m)/sizeof(m[0]); ++i )
-+ if ( m[i].value == result ) return m[i].message;
-+
-+ return "Unknown error.";
-+}
-+
-+//******************** End of __WINDOWS_ASIO__ *********************//
-+#endif
-+
-+
-+#if defined(__WINDOWS_WASAPI__) // Windows WASAPI API
-+
-+// Authored by Marcus Tomlinson <themarcustomlinson@gmail.com>, April 2014
-+// - Introduces support for the Windows WASAPI API
-+// - Aims to deliver bit streams to and from hardware at the lowest possible latency, via the absolute minimum buffer sizes required
-+// - Provides flexible stream configuration to an otherwise strict and inflexible WASAPI interface
-+// - Includes automatic internal conversion of sample rate and buffer size between hardware and the user
-+
-+#ifndef INITGUID
-+ #define INITGUID
-+#endif
-+#include <audioclient.h>
-+#include <avrt.h>
-+#include <mmdeviceapi.h>
-+#include <functiondiscoverykeys_devpkey.h>
-+
-+//=============================================================================
-+
-+#define SAFE_RELEASE( objectPtr )\
-+if ( objectPtr )\
-+{\
-+ objectPtr->Release();\
-+ objectPtr = NULL;\
-+}
-+
-+typedef HANDLE ( __stdcall *TAvSetMmThreadCharacteristicsPtr )( LPCWSTR TaskName, LPDWORD TaskIndex );
-+
-+//-----------------------------------------------------------------------------
-+
-+// WASAPI dictates stream sample rate, format, channel count, and in some cases, buffer size.
-+// Therefore we must perform all necessary conversions to user buffers in order to satisfy these
-+// requirements. WasapiBuffer ring buffers are used between HwIn->UserIn and UserOut->HwOut to
-+// provide intermediate storage for read / write synchronization.
-+class WasapiBuffer
-+{
-+public:
-+ WasapiBuffer()
-+ : buffer_( NULL ),
-+ bufferSize_( 0 ),
-+ inIndex_( 0 ),
-+ outIndex_( 0 ) {}
-+
-+ ~WasapiBuffer() {
-+ free( buffer_ );
-+ }
-+
-+ // sets the length of the internal ring buffer
-+ void setBufferSize( unsigned int bufferSize, unsigned int formatBytes ) {
-+ free( buffer_ );
-+
-+ buffer_ = ( char* ) calloc( bufferSize, formatBytes );
-+
-+ bufferSize_ = bufferSize;
-+ inIndex_ = 0;
-+ outIndex_ = 0;
-+ }
-+
-+ // attempt to push a buffer into the ring buffer at the current "in" index
-+ bool pushBuffer( char* buffer, unsigned int bufferSize, RtAudioFormat format )
-+ {
-+ if ( !buffer || // incoming buffer is NULL
-+ bufferSize == 0 || // incoming buffer has no data
-+ bufferSize > bufferSize_ ) // incoming buffer too large
-+ {
-+ return false;
-+ }
-+
-+ unsigned int relOutIndex = outIndex_;
-+ unsigned int inIndexEnd = inIndex_ + bufferSize;
-+ if ( relOutIndex < inIndex_ && inIndexEnd >= bufferSize_ ) {
-+ relOutIndex += bufferSize_;
-+ }
-+
-+ // "in" index can end on the "out" index but cannot begin at it
-+ if ( inIndex_ <= relOutIndex && inIndexEnd > relOutIndex ) {
-+ return false; // not enough space between "in" index and "out" index
-+ }
-+
-+ // copy buffer from external to internal
-+ int fromZeroSize = inIndex_ + bufferSize - bufferSize_;
-+ fromZeroSize = fromZeroSize < 0 ? 0 : fromZeroSize;
-+ int fromInSize = bufferSize - fromZeroSize;
-+
-+ switch( format )
-+ {
-+ case RTAUDIO_SINT8:
-+ memcpy( &( ( char* ) buffer_ )[inIndex_], buffer, fromInSize * sizeof( char ) );
-+ memcpy( buffer_, &( ( char* ) buffer )[fromInSize], fromZeroSize * sizeof( char ) );
-+ break;
-+ case RTAUDIO_SINT16:
-+ memcpy( &( ( short* ) buffer_ )[inIndex_], buffer, fromInSize * sizeof( short ) );
-+ memcpy( buffer_, &( ( short* ) buffer )[fromInSize], fromZeroSize * sizeof( short ) );
-+ break;
-+ case RTAUDIO_SINT24:
-+ memcpy( &( ( S24* ) buffer_ )[inIndex_], buffer, fromInSize * sizeof( S24 ) );
-+ memcpy( buffer_, &( ( S24* ) buffer )[fromInSize], fromZeroSize * sizeof( S24 ) );
-+ break;
-+ case RTAUDIO_SINT32:
-+ memcpy( &( ( int* ) buffer_ )[inIndex_], buffer, fromInSize * sizeof( int ) );
-+ memcpy( buffer_, &( ( int* ) buffer )[fromInSize], fromZeroSize * sizeof( int ) );
-+ break;
-+ case RTAUDIO_FLOAT32:
-+ memcpy( &( ( float* ) buffer_ )[inIndex_], buffer, fromInSize * sizeof( float ) );
-+ memcpy( buffer_, &( ( float* ) buffer )[fromInSize], fromZeroSize * sizeof( float ) );
-+ break;
-+ case RTAUDIO_FLOAT64:
-+ memcpy( &( ( double* ) buffer_ )[inIndex_], buffer, fromInSize * sizeof( double ) );
-+ memcpy( buffer_, &( ( double* ) buffer )[fromInSize], fromZeroSize * sizeof( double ) );
-+ break;
-+ }
-+
-+ // update "in" index
-+ inIndex_ += bufferSize;
-+ inIndex_ %= bufferSize_;
-+
-+ return true;
-+ }
-+
-+ // attempt to pull a buffer from the ring buffer from the current "out" index
-+ bool pullBuffer( char* buffer, unsigned int bufferSize, RtAudioFormat format )
-+ {
-+ if ( !buffer || // incoming buffer is NULL
-+ bufferSize == 0 || // incoming buffer has no data
-+ bufferSize > bufferSize_ ) // incoming buffer too large
-+ {
-+ return false;
-+ }
-+
-+ unsigned int relInIndex = inIndex_;
-+ unsigned int outIndexEnd = outIndex_ + bufferSize;
-+ if ( relInIndex < outIndex_ && outIndexEnd >= bufferSize_ ) {
-+ relInIndex += bufferSize_;
-+ }
-+
-+ // "out" index can begin at and end on the "in" index
-+ if ( outIndex_ < relInIndex && outIndexEnd > relInIndex ) {
-+ return false; // not enough space between "out" index and "in" index
-+ }
-+
-+ // copy buffer from internal to external
-+ int fromZeroSize = outIndex_ + bufferSize - bufferSize_;
-+ fromZeroSize = fromZeroSize < 0 ? 0 : fromZeroSize;
-+ int fromOutSize = bufferSize - fromZeroSize;
-+
-+ switch( format )
-+ {
-+ case RTAUDIO_SINT8:
-+ memcpy( buffer, &( ( char* ) buffer_ )[outIndex_], fromOutSize * sizeof( char ) );
-+ memcpy( &( ( char* ) buffer )[fromOutSize], buffer_, fromZeroSize * sizeof( char ) );
-+ break;
-+ case RTAUDIO_SINT16:
-+ memcpy( buffer, &( ( short* ) buffer_ )[outIndex_], fromOutSize * sizeof( short ) );
-+ memcpy( &( ( short* ) buffer )[fromOutSize], buffer_, fromZeroSize * sizeof( short ) );
-+ break;
-+ case RTAUDIO_SINT24:
-+ memcpy( buffer, &( ( S24* ) buffer_ )[outIndex_], fromOutSize * sizeof( S24 ) );
-+ memcpy( &( ( S24* ) buffer )[fromOutSize], buffer_, fromZeroSize * sizeof( S24 ) );
-+ break;
-+ case RTAUDIO_SINT32:
-+ memcpy( buffer, &( ( int* ) buffer_ )[outIndex_], fromOutSize * sizeof( int ) );
-+ memcpy( &( ( int* ) buffer )[fromOutSize], buffer_, fromZeroSize * sizeof( int ) );
-+ break;
-+ case RTAUDIO_FLOAT32:
-+ memcpy( buffer, &( ( float* ) buffer_ )[outIndex_], fromOutSize * sizeof( float ) );
-+ memcpy( &( ( float* ) buffer )[fromOutSize], buffer_, fromZeroSize * sizeof( float ) );
-+ break;
-+ case RTAUDIO_FLOAT64:
-+ memcpy( buffer, &( ( double* ) buffer_ )[outIndex_], fromOutSize * sizeof( double ) );
-+ memcpy( &( ( double* ) buffer )[fromOutSize], buffer_, fromZeroSize * sizeof( double ) );
-+ break;
-+ }
-+
-+ // update "out" index
-+ outIndex_ += bufferSize;
-+ outIndex_ %= bufferSize_;
-+
-+ return true;
-+ }
-+
-+private:
-+ char* buffer_;
-+ unsigned int bufferSize_;
-+ unsigned int inIndex_;
-+ unsigned int outIndex_;
-+};
-+
-+//-----------------------------------------------------------------------------
-+
-+// In order to satisfy WASAPI's buffer requirements, we need a means of converting sample rate
-+// between HW and the user. The convertBufferWasapi function is used to perform this conversion
-+// between HwIn->UserIn and UserOut->HwOut during the stream callback loop.
-+// This sample rate converter works best with conversions between one rate and its multiple.
-+void convertBufferWasapi( char* outBuffer,
-+ const char* inBuffer,
-+ const unsigned int& channelCount,
-+ const unsigned int& inSampleRate,
-+ const unsigned int& outSampleRate,
-+ const unsigned int& inSampleCount,
-+ unsigned int& outSampleCount,
-+ const RtAudioFormat& format )
-+{
-+ // calculate the new outSampleCount and relative sampleStep
-+ float sampleRatio = ( float ) outSampleRate / inSampleRate;
-+ float sampleRatioInv = ( float ) 1 / sampleRatio;
-+ float sampleStep = 1.0f / sampleRatio;
-+ float inSampleFraction = 0.0f;
-+
-+ outSampleCount = ( unsigned int ) std::roundf( inSampleCount * sampleRatio );
-+
-+ // if inSampleRate is a multiple of outSampleRate (or vice versa) there's no need to interpolate
-+ if ( floor( sampleRatio ) == sampleRatio || floor( sampleRatioInv ) == sampleRatioInv )
-+ {
-+ // frame-by-frame, copy each relative input sample into it's corresponding output sample
-+ for ( unsigned int outSample = 0; outSample < outSampleCount; outSample++ )
-+ {
-+ unsigned int inSample = ( unsigned int ) inSampleFraction;
-+
-+ switch ( format )
-+ {
-+ case RTAUDIO_SINT8:
-+ memcpy( &( ( char* ) outBuffer )[ outSample * channelCount ], &( ( char* ) inBuffer )[ inSample * channelCount ], channelCount * sizeof( char ) );
-+ break;
-+ case RTAUDIO_SINT16:
-+ memcpy( &( ( short* ) outBuffer )[ outSample * channelCount ], &( ( short* ) inBuffer )[ inSample * channelCount ], channelCount * sizeof( short ) );
-+ break;
-+ case RTAUDIO_SINT24:
-+ memcpy( &( ( S24* ) outBuffer )[ outSample * channelCount ], &( ( S24* ) inBuffer )[ inSample * channelCount ], channelCount * sizeof( S24 ) );
-+ break;
-+ case RTAUDIO_SINT32:
-+ memcpy( &( ( int* ) outBuffer )[ outSample * channelCount ], &( ( int* ) inBuffer )[ inSample * channelCount ], channelCount * sizeof( int ) );
-+ break;
-+ case RTAUDIO_FLOAT32:
-+ memcpy( &( ( float* ) outBuffer )[ outSample * channelCount ], &( ( float* ) inBuffer )[ inSample * channelCount ], channelCount * sizeof( float ) );
-+ break;
-+ case RTAUDIO_FLOAT64:
-+ memcpy( &( ( double* ) outBuffer )[ outSample * channelCount ], &( ( double* ) inBuffer )[ inSample * channelCount ], channelCount * sizeof( double ) );
-+ break;
-+ }
-+
-+ // jump to next in sample
-+ inSampleFraction += sampleStep;
-+ }
-+ }
-+ else // else interpolate
-+ {
-+ // frame-by-frame, copy each relative input sample into it's corresponding output sample
-+ for ( unsigned int outSample = 0; outSample < outSampleCount; outSample++ )
-+ {
-+ unsigned int inSample = ( unsigned int ) inSampleFraction;
-+ float inSampleDec = inSampleFraction - inSample;
-+ unsigned int frameInSample = inSample * channelCount;
-+ unsigned int frameOutSample = outSample * channelCount;
-+
-+ switch ( format )
-+ {
-+ case RTAUDIO_SINT8:
-+ {
-+ for ( unsigned int channel = 0; channel < channelCount; channel++ )
-+ {
-+ char fromSample = ( ( char* ) inBuffer )[ frameInSample + channel ];
-+ char toSample = ( ( char* ) inBuffer )[ frameInSample + channelCount + channel ];
-+ char sampleDiff = ( char ) ( ( toSample - fromSample ) * inSampleDec );
-+ ( ( char* ) outBuffer )[ frameOutSample + channel ] = fromSample + sampleDiff;
-+ }
-+ break;
-+ }
-+ case RTAUDIO_SINT16:
-+ {
-+ for ( unsigned int channel = 0; channel < channelCount; channel++ )
-+ {
-+ short fromSample = ( ( short* ) inBuffer )[ frameInSample + channel ];
-+ short toSample = ( ( short* ) inBuffer )[ frameInSample + channelCount + channel ];
-+ short sampleDiff = ( short ) ( ( toSample - fromSample ) * inSampleDec );
-+ ( ( short* ) outBuffer )[ frameOutSample + channel ] = fromSample + sampleDiff;
-+ }
-+ break;
-+ }
-+ case RTAUDIO_SINT24:
-+ {
-+ for ( unsigned int channel = 0; channel < channelCount; channel++ )
-+ {
-+ int fromSample = ( ( S24* ) inBuffer )[ frameInSample + channel ].asInt();
-+ int toSample = ( ( S24* ) inBuffer )[ frameInSample + channelCount + channel ].asInt();
-+ int sampleDiff = ( int ) ( ( toSample - fromSample ) * inSampleDec );
-+ ( ( S24* ) outBuffer )[ frameOutSample + channel ] = fromSample + sampleDiff;
-+ }
-+ break;
-+ }
-+ case RTAUDIO_SINT32:
-+ {
-+ for ( unsigned int channel = 0; channel < channelCount; channel++ )
-+ {
-+ int fromSample = ( ( int* ) inBuffer )[ frameInSample + channel ];
-+ int toSample = ( ( int* ) inBuffer )[ frameInSample + channelCount + channel ];
-+ int sampleDiff = ( int ) ( ( toSample - fromSample ) * inSampleDec );
-+ ( ( int* ) outBuffer )[ frameOutSample + channel ] = fromSample + sampleDiff;
-+ }
-+ break;
-+ }
-+ case RTAUDIO_FLOAT32:
-+ {
-+ for ( unsigned int channel = 0; channel < channelCount; channel++ )
-+ {
-+ float fromSample = ( ( float* ) inBuffer )[ frameInSample + channel ];
-+ float toSample = ( ( float* ) inBuffer )[ frameInSample + channelCount + channel ];
-+ float sampleDiff = ( toSample - fromSample ) * inSampleDec;
-+ ( ( float* ) outBuffer )[ frameOutSample + channel ] = fromSample + sampleDiff;
-+ }
-+ break;
-+ }
-+ case RTAUDIO_FLOAT64:
-+ {
-+ for ( unsigned int channel = 0; channel < channelCount; channel++ )
-+ {
-+ double fromSample = ( ( double* ) inBuffer )[ frameInSample + channel ];
-+ double toSample = ( ( double* ) inBuffer )[ frameInSample + channelCount + channel ];
-+ double sampleDiff = ( toSample - fromSample ) * inSampleDec;
-+ ( ( double* ) outBuffer )[ frameOutSample + channel ] = fromSample + sampleDiff;
-+ }
-+ break;
-+ }
-+ }
-+
-+ // jump to next in sample
-+ inSampleFraction += sampleStep;
-+ }
-+ }
-+}
-+
-+//-----------------------------------------------------------------------------
-+
-+// A structure to hold various information related to the WASAPI implementation.
-+struct WasapiHandle
-+{
-+ IAudioClient* captureAudioClient;
-+ IAudioClient* renderAudioClient;
-+ IAudioCaptureClient* captureClient;
-+ IAudioRenderClient* renderClient;
-+ HANDLE captureEvent;
-+ HANDLE renderEvent;
-+
-+ WasapiHandle()
-+ : captureAudioClient( NULL ),
-+ renderAudioClient( NULL ),
-+ captureClient( NULL ),
-+ renderClient( NULL ),
-+ captureEvent( NULL ),
-+ renderEvent( NULL ) {}
-+};
-+
-+//=============================================================================
-+
-+RtApiWasapi::RtApiWasapi()
-+ : coInitialized_( false ), deviceEnumerator_( NULL )
-+{
-+ // WASAPI can run either apartment or multi-threaded
-+ HRESULT hr = CoInitialize( NULL );
-+ if ( !FAILED( hr ) )
-+ coInitialized_ = true;
-+
-+ // Instantiate device enumerator
-+ hr = CoCreateInstance( __uuidof( MMDeviceEnumerator ), NULL,
-+ CLSCTX_ALL, __uuidof( IMMDeviceEnumerator ),
-+ ( void** ) &deviceEnumerator_ );
-+
-+ if ( FAILED( hr ) ) {
-+ errorText_ = "RtApiWasapi::RtApiWasapi: Unable to instantiate device enumerator";
-+ error( RtAudioError::DRIVER_ERROR );
-+ }
-+}
-+
-+//-----------------------------------------------------------------------------
-+
-+RtApiWasapi::~RtApiWasapi()
-+{
-+ if ( stream_.state != STREAM_CLOSED )
-+ closeStream();
-+
-+ SAFE_RELEASE( deviceEnumerator_ );
-+
-+ // If this object previously called CoInitialize()
-+ if ( coInitialized_ )
-+ CoUninitialize();
-+}
-+
-+//=============================================================================
-+
-+unsigned int RtApiWasapi::getDeviceCount( void )
-+{
-+ unsigned int captureDeviceCount = 0;
-+ unsigned int renderDeviceCount = 0;
-+
-+ IMMDeviceCollection* captureDevices = NULL;
-+ IMMDeviceCollection* renderDevices = NULL;
-+
-+ // Count capture devices
-+ errorText_.clear();
-+ HRESULT hr = deviceEnumerator_->EnumAudioEndpoints( eCapture, DEVICE_STATE_ACTIVE, &captureDevices );
-+ if ( FAILED( hr ) ) {
-+ errorText_ = "RtApiWasapi::getDeviceCount: Unable to retrieve capture device collection.";
-+ goto Exit;
-+ }
-+
-+ hr = captureDevices->GetCount( &captureDeviceCount );
-+ if ( FAILED( hr ) ) {
-+ errorText_ = "RtApiWasapi::getDeviceCount: Unable to retrieve capture device count.";
-+ goto Exit;
-+ }
-+
-+ // Count render devices
-+ hr = deviceEnumerator_->EnumAudioEndpoints( eRender, DEVICE_STATE_ACTIVE, &renderDevices );
-+ if ( FAILED( hr ) ) {
-+ errorText_ = "RtApiWasapi::getDeviceCount: Unable to retrieve render device collection.";
-+ goto Exit;
-+ }
-+
-+ hr = renderDevices->GetCount( &renderDeviceCount );
-+ if ( FAILED( hr ) ) {
-+ errorText_ = "RtApiWasapi::getDeviceCount: Unable to retrieve render device count.";
-+ goto Exit;
-+ }
-+
-+Exit:
-+ // release all references
-+ SAFE_RELEASE( captureDevices );
-+ SAFE_RELEASE( renderDevices );
-+
-+ if ( errorText_.empty() )
-+ return captureDeviceCount + renderDeviceCount;
-+
-+ error( RtAudioError::DRIVER_ERROR );
-+ return 0;
-+}
-+
-+//-----------------------------------------------------------------------------
-+
-+RtAudio::DeviceInfo RtApiWasapi::getDeviceInfo( unsigned int device )
-+{
-+ RtAudio::DeviceInfo info;
-+ unsigned int captureDeviceCount = 0;
-+ unsigned int renderDeviceCount = 0;
-+ std::string defaultDeviceName;
-+ bool isCaptureDevice = false;
-+
-+ PROPVARIANT deviceNameProp;
-+ PROPVARIANT defaultDeviceNameProp;
-+
-+ IMMDeviceCollection* captureDevices = NULL;
-+ IMMDeviceCollection* renderDevices = NULL;
-+ IMMDevice* devicePtr = NULL;
-+ IMMDevice* defaultDevicePtr = NULL;
-+ IAudioClient* audioClient = NULL;
-+ IPropertyStore* devicePropStore = NULL;
-+ IPropertyStore* defaultDevicePropStore = NULL;
-+
-+ WAVEFORMATEX* deviceFormat = NULL;
-+ WAVEFORMATEX* closestMatchFormat = NULL;
-+
-+ // probed
-+ info.probed = false;
-+
-+ // Count capture devices
-+ errorText_.clear();
-+ RtAudioError::Type errorType = RtAudioError::DRIVER_ERROR;
-+ HRESULT hr = deviceEnumerator_->EnumAudioEndpoints( eCapture, DEVICE_STATE_ACTIVE, &captureDevices );
-+ if ( FAILED( hr ) ) {
-+ errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve capture device collection.";
-+ goto Exit;
-+ }
-+
-+ hr = captureDevices->GetCount( &captureDeviceCount );
-+ if ( FAILED( hr ) ) {
-+ errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve capture device count.";
-+ goto Exit;
-+ }
-+
-+ // Count render devices
-+ hr = deviceEnumerator_->EnumAudioEndpoints( eRender, DEVICE_STATE_ACTIVE, &renderDevices );
-+ if ( FAILED( hr ) ) {
-+ errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve render device collection.";
-+ goto Exit;
-+ }
-+
-+ hr = renderDevices->GetCount( &renderDeviceCount );
-+ if ( FAILED( hr ) ) {
-+ errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve render device count.";
-+ goto Exit;
-+ }
-+
-+ // validate device index
-+ if ( device >= captureDeviceCount + renderDeviceCount ) {
-+ errorText_ = "RtApiWasapi::getDeviceInfo: Invalid device index.";
-+ errorType = RtAudioError::INVALID_USE;
-+ goto Exit;
-+ }
-+
-+ // determine whether index falls within capture or render devices
-+ if ( device >= renderDeviceCount ) {
-+ hr = captureDevices->Item( device - renderDeviceCount, &devicePtr );
-+ if ( FAILED( hr ) ) {
-+ errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve capture device handle.";
-+ goto Exit;
-+ }
-+ isCaptureDevice = true;
-+ }
-+ else {
-+ hr = renderDevices->Item( device, &devicePtr );
-+ if ( FAILED( hr ) ) {
-+ errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve render device handle.";
-+ goto Exit;
-+ }
-+ isCaptureDevice = false;
-+ }
-+
-+ // get default device name
-+ if ( isCaptureDevice ) {
-+ hr = deviceEnumerator_->GetDefaultAudioEndpoint( eCapture, eConsole, &defaultDevicePtr );
-+ if ( FAILED( hr ) ) {
-+ errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve default capture device handle.";
-+ goto Exit;
-+ }
-+ }
-+ else {
-+ hr = deviceEnumerator_->GetDefaultAudioEndpoint( eRender, eConsole, &defaultDevicePtr );
-+ if ( FAILED( hr ) ) {
-+ errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve default render device handle.";
-+ goto Exit;
-+ }
-+ }
-+
-+ hr = defaultDevicePtr->OpenPropertyStore( STGM_READ, &defaultDevicePropStore );
-+ if ( FAILED( hr ) ) {
-+ errorText_ = "RtApiWasapi::getDeviceInfo: Unable to open default device property store.";
-+ goto Exit;
-+ }
-+ PropVariantInit( &defaultDeviceNameProp );
-+
-+ hr = defaultDevicePropStore->GetValue( PKEY_Device_FriendlyName, &defaultDeviceNameProp );
-+ if ( FAILED( hr ) ) {
-+ errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve default device property: PKEY_Device_FriendlyName.";
-+ goto Exit;
-+ }
-+
-+ defaultDeviceName = convertCharPointerToStdString(defaultDeviceNameProp.pwszVal);
-+
-+ // name
-+ hr = devicePtr->OpenPropertyStore( STGM_READ, &devicePropStore );
-+ if ( FAILED( hr ) ) {
-+ errorText_ = "RtApiWasapi::getDeviceInfo: Unable to open device property store.";
-+ goto Exit;
-+ }
-+
-+ PropVariantInit( &deviceNameProp );
-+
-+ hr = devicePropStore->GetValue( PKEY_Device_FriendlyName, &deviceNameProp );
-+ if ( FAILED( hr ) ) {
-+ errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve device property: PKEY_Device_FriendlyName.";
-+ goto Exit;
-+ }
-+
-+ info.name =convertCharPointerToStdString(deviceNameProp.pwszVal);
-+
-+ // is default
-+ if ( isCaptureDevice ) {
-+ info.isDefaultInput = info.name == defaultDeviceName;
-+ info.isDefaultOutput = false;
-+ }
-+ else {
-+ info.isDefaultInput = false;
-+ info.isDefaultOutput = info.name == defaultDeviceName;
-+ }
-+
-+ // channel count
-+ hr = devicePtr->Activate( __uuidof( IAudioClient ), CLSCTX_ALL, NULL, ( void** ) &audioClient );
-+ if ( FAILED( hr ) ) {
-+ errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve device audio client.";
-+ goto Exit;
-+ }
-+
-+ hr = audioClient->GetMixFormat( &deviceFormat );
-+ if ( FAILED( hr ) ) {
-+ errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve device mix format.";
-+ goto Exit;
-+ }
-+
-+ if ( isCaptureDevice ) {
-+ info.inputChannels = deviceFormat->nChannels;
-+ info.outputChannels = 0;
-+ info.duplexChannels = 0;
-+ }
-+ else {
-+ info.inputChannels = 0;
-+ info.outputChannels = deviceFormat->nChannels;
-+ info.duplexChannels = 0;
-+ }
-+
-+ // sample rates
-+ info.sampleRates.clear();
-+
-+ // allow support for all sample rates as we have a built-in sample rate converter
-+ for ( unsigned int i = 0; i < MAX_SAMPLE_RATES; i++ ) {
-+ info.sampleRates.push_back( SAMPLE_RATES[i] );
-+ }
-+ info.preferredSampleRate = deviceFormat->nSamplesPerSec;
-+
-+ // native format
-+ info.nativeFormats = 0;
-+
-+ if ( deviceFormat->wFormatTag == WAVE_FORMAT_IEEE_FLOAT ||
-+ ( deviceFormat->wFormatTag == WAVE_FORMAT_EXTENSIBLE &&
-+ ( ( WAVEFORMATEXTENSIBLE* ) deviceFormat )->SubFormat == KSDATAFORMAT_SUBTYPE_IEEE_FLOAT ) )
-+ {
-+ if ( deviceFormat->wBitsPerSample == 32 ) {
-+ info.nativeFormats |= RTAUDIO_FLOAT32;
-+ }
-+ else if ( deviceFormat->wBitsPerSample == 64 ) {
-+ info.nativeFormats |= RTAUDIO_FLOAT64;
-+ }
-+ }
-+ else if ( deviceFormat->wFormatTag == WAVE_FORMAT_PCM ||
-+ ( deviceFormat->wFormatTag == WAVE_FORMAT_EXTENSIBLE &&
-+ ( ( WAVEFORMATEXTENSIBLE* ) deviceFormat )->SubFormat == KSDATAFORMAT_SUBTYPE_PCM ) )
-+ {
-+ if ( deviceFormat->wBitsPerSample == 8 ) {
-+ info.nativeFormats |= RTAUDIO_SINT8;
-+ }
-+ else if ( deviceFormat->wBitsPerSample == 16 ) {
-+ info.nativeFormats |= RTAUDIO_SINT16;
-+ }
-+ else if ( deviceFormat->wBitsPerSample == 24 ) {
-+ info.nativeFormats |= RTAUDIO_SINT24;
-+ }
-+ else if ( deviceFormat->wBitsPerSample == 32 ) {
-+ info.nativeFormats |= RTAUDIO_SINT32;
-+ }
-+ }
-+
-+ // probed
-+ info.probed = true;
-+
-+Exit:
-+ // release all references
-+ PropVariantClear( &deviceNameProp );
-+ PropVariantClear( &defaultDeviceNameProp );
-+
-+ SAFE_RELEASE( captureDevices );
-+ SAFE_RELEASE( renderDevices );
-+ SAFE_RELEASE( devicePtr );
-+ SAFE_RELEASE( defaultDevicePtr );
-+ SAFE_RELEASE( audioClient );
-+ SAFE_RELEASE( devicePropStore );
-+ SAFE_RELEASE( defaultDevicePropStore );
-+
-+ CoTaskMemFree( deviceFormat );
-+ CoTaskMemFree( closestMatchFormat );
-+
-+ if ( !errorText_.empty() )
-+ error( errorType );
-+ return info;
-+}
-+
-+//-----------------------------------------------------------------------------
-+
-+unsigned int RtApiWasapi::getDefaultOutputDevice( void )
-+{
-+ for ( unsigned int i = 0; i < getDeviceCount(); i++ ) {
-+ if ( getDeviceInfo( i ).isDefaultOutput ) {
-+ return i;
-+ }
-+ }
-+
-+ return 0;
-+}
-+
-+//-----------------------------------------------------------------------------
-+
-+unsigned int RtApiWasapi::getDefaultInputDevice( void )
-+{
-+ for ( unsigned int i = 0; i < getDeviceCount(); i++ ) {
-+ if ( getDeviceInfo( i ).isDefaultInput ) {
-+ return i;
-+ }
-+ }
-+
-+ return 0;
-+}
-+
-+//-----------------------------------------------------------------------------
-+
-+void RtApiWasapi::closeStream( void )
-+{
-+ if ( stream_.state == STREAM_CLOSED ) {
-+ errorText_ = "RtApiWasapi::closeStream: No open stream to close.";
-+ error( RtAudioError::WARNING );
-+ return;
-+ }
-+
-+ if ( stream_.state != STREAM_STOPPED )
-+ stopStream();
-+
-+ // clean up stream memory
-+ SAFE_RELEASE( ( ( WasapiHandle* ) stream_.apiHandle )->captureAudioClient )
-+ SAFE_RELEASE( ( ( WasapiHandle* ) stream_.apiHandle )->renderAudioClient )
-+
-+ SAFE_RELEASE( ( ( WasapiHandle* ) stream_.apiHandle )->captureClient )
-+ SAFE_RELEASE( ( ( WasapiHandle* ) stream_.apiHandle )->renderClient )
-+
-+ if ( ( ( WasapiHandle* ) stream_.apiHandle )->captureEvent )
-+ CloseHandle( ( ( WasapiHandle* ) stream_.apiHandle )->captureEvent );
-+
-+ if ( ( ( WasapiHandle* ) stream_.apiHandle )->renderEvent )
-+ CloseHandle( ( ( WasapiHandle* ) stream_.apiHandle )->renderEvent );
-+
-+ delete ( WasapiHandle* ) stream_.apiHandle;
-+ stream_.apiHandle = NULL;
-+
-+ for ( int i = 0; i < 2; i++ ) {
-+ if ( stream_.userBuffer[i] ) {
-+ free( stream_.userBuffer[i] );
-+ stream_.userBuffer[i] = 0;
-+ }
-+ }
-+
-+ if ( stream_.deviceBuffer ) {
-+ free( stream_.deviceBuffer );
-+ stream_.deviceBuffer = 0;
-+ }
-+
-+ // update stream state
-+ stream_.state = STREAM_CLOSED;
-+}
-+
-+//-----------------------------------------------------------------------------
-+
-+void RtApiWasapi::startStream( void )
-+{
-+ verifyStream();
-+
-+ if ( stream_.state == STREAM_RUNNING ) {
-+ errorText_ = "RtApiWasapi::startStream: The stream is already running.";
-+ error( RtAudioError::WARNING );
-+ return;
-+ }
-+
-+ // update stream state
-+ stream_.state = STREAM_RUNNING;
-+
-+ // create WASAPI stream thread
-+ stream_.callbackInfo.thread = ( ThreadHandle ) CreateThread( NULL, 0, runWasapiThread, this, CREATE_SUSPENDED, NULL );
-+
-+ if ( !stream_.callbackInfo.thread ) {
-+ errorText_ = "RtApiWasapi::startStream: Unable to instantiate callback thread.";
-+ error( RtAudioError::THREAD_ERROR );
-+ }
-+ else {
-+ SetThreadPriority( ( void* ) stream_.callbackInfo.thread, stream_.callbackInfo.priority );
-+ ResumeThread( ( void* ) stream_.callbackInfo.thread );
-+ }
-+}
-+
-+//-----------------------------------------------------------------------------
-+
-+void RtApiWasapi::stopStream( void )
-+{
-+ verifyStream();
-+
-+ if ( stream_.state == STREAM_STOPPED ) {
-+ errorText_ = "RtApiWasapi::stopStream: The stream is already stopped.";
-+ error( RtAudioError::WARNING );
-+ return;
-+ }
-+
-+ // inform stream thread by setting stream state to STREAM_STOPPING
-+ stream_.state = STREAM_STOPPING;
-+
-+ // wait until stream thread is stopped
-+ while( stream_.state != STREAM_STOPPED ) {
-+ Sleep( 1 );
-+ }
-+
-+ // Wait for the last buffer to play before stopping.
-+ Sleep( 1000 * stream_.bufferSize / stream_.sampleRate );
-+
-+ // stop capture client if applicable
-+ if ( ( ( WasapiHandle* ) stream_.apiHandle )->captureAudioClient ) {
-+ HRESULT hr = ( ( WasapiHandle* ) stream_.apiHandle )->captureAudioClient->Stop();
-+ if ( FAILED( hr ) ) {
-+ errorText_ = "RtApiWasapi::stopStream: Unable to stop capture stream.";
-+ error( RtAudioError::DRIVER_ERROR );
-+ return;
-+ }
-+ }
-+
-+ // stop render client if applicable
-+ if ( ( ( WasapiHandle* ) stream_.apiHandle )->renderAudioClient ) {
-+ HRESULT hr = ( ( WasapiHandle* ) stream_.apiHandle )->renderAudioClient->Stop();
-+ if ( FAILED( hr ) ) {
-+ errorText_ = "RtApiWasapi::stopStream: Unable to stop render stream.";
-+ error( RtAudioError::DRIVER_ERROR );
-+ return;
-+ }
-+ }
-+
-+ // close thread handle
-+ if ( stream_.callbackInfo.thread && !CloseHandle( ( void* ) stream_.callbackInfo.thread ) ) {
-+ errorText_ = "RtApiWasapi::stopStream: Unable to close callback thread.";
-+ error( RtAudioError::THREAD_ERROR );
-+ return;
-+ }
-+
-+ stream_.callbackInfo.thread = (ThreadHandle) NULL;
-+}
-+
-+//-----------------------------------------------------------------------------
-+
-+void RtApiWasapi::abortStream( void )
-+{
-+ verifyStream();
-+
-+ if ( stream_.state == STREAM_STOPPED ) {
-+ errorText_ = "RtApiWasapi::abortStream: The stream is already stopped.";
-+ error( RtAudioError::WARNING );
-+ return;
-+ }
-+
-+ // inform stream thread by setting stream state to STREAM_STOPPING
-+ stream_.state = STREAM_STOPPING;
-+
-+ // wait until stream thread is stopped
-+ while ( stream_.state != STREAM_STOPPED ) {
-+ Sleep( 1 );
-+ }
-+
-+ // stop capture client if applicable
-+ if ( ( ( WasapiHandle* ) stream_.apiHandle )->captureAudioClient ) {
-+ HRESULT hr = ( ( WasapiHandle* ) stream_.apiHandle )->captureAudioClient->Stop();
-+ if ( FAILED( hr ) ) {
-+ errorText_ = "RtApiWasapi::abortStream: Unable to stop capture stream.";
-+ error( RtAudioError::DRIVER_ERROR );
-+ return;
-+ }
-+ }
-+
-+ // stop render client if applicable
-+ if ( ( ( WasapiHandle* ) stream_.apiHandle )->renderAudioClient ) {
-+ HRESULT hr = ( ( WasapiHandle* ) stream_.apiHandle )->renderAudioClient->Stop();
-+ if ( FAILED( hr ) ) {
-+ errorText_ = "RtApiWasapi::abortStream: Unable to stop render stream.";
-+ error( RtAudioError::DRIVER_ERROR );
-+ return;
-+ }
-+ }
-+
-+ // close thread handle
-+ if ( stream_.callbackInfo.thread && !CloseHandle( ( void* ) stream_.callbackInfo.thread ) ) {
-+ errorText_ = "RtApiWasapi::abortStream: Unable to close callback thread.";
-+ error( RtAudioError::THREAD_ERROR );
-+ return;
-+ }
-+
-+ stream_.callbackInfo.thread = (ThreadHandle) NULL;
-+}
-+
-+//-----------------------------------------------------------------------------
-+
-+bool RtApiWasapi::probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,
-+ unsigned int firstChannel, unsigned int sampleRate,
-+ RtAudioFormat format, unsigned int* bufferSize,
-+ RtAudio::StreamOptions* options )
-+{
-+ bool methodResult = FAILURE;
-+ unsigned int captureDeviceCount = 0;
-+ unsigned int renderDeviceCount = 0;
-+
-+ IMMDeviceCollection* captureDevices = NULL;
-+ IMMDeviceCollection* renderDevices = NULL;
-+ IMMDevice* devicePtr = NULL;
-+ WAVEFORMATEX* deviceFormat = NULL;
-+ unsigned int bufferBytes;
-+ stream_.state = STREAM_STOPPED;
-+
-+ // create API Handle if not already created
-+ if ( !stream_.apiHandle )
-+ stream_.apiHandle = ( void* ) new WasapiHandle();
-+
-+ // Count capture devices
-+ errorText_.clear();
-+ RtAudioError::Type errorType = RtAudioError::DRIVER_ERROR;
-+ HRESULT hr = deviceEnumerator_->EnumAudioEndpoints( eCapture, DEVICE_STATE_ACTIVE, &captureDevices );
-+ if ( FAILED( hr ) ) {
-+ errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve capture device collection.";
-+ goto Exit;
-+ }
-+
-+ hr = captureDevices->GetCount( &captureDeviceCount );
-+ if ( FAILED( hr ) ) {
-+ errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve capture device count.";
-+ goto Exit;
-+ }
-+
-+ // Count render devices
-+ hr = deviceEnumerator_->EnumAudioEndpoints( eRender, DEVICE_STATE_ACTIVE, &renderDevices );
-+ if ( FAILED( hr ) ) {
-+ errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve render device collection.";
-+ goto Exit;
-+ }
-+
-+ hr = renderDevices->GetCount( &renderDeviceCount );
-+ if ( FAILED( hr ) ) {
-+ errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve render device count.";
-+ goto Exit;
-+ }
-+
-+ // validate device index
-+ if ( device >= captureDeviceCount + renderDeviceCount ) {
-+ errorType = RtAudioError::INVALID_USE;
-+ errorText_ = "RtApiWasapi::probeDeviceOpen: Invalid device index.";
-+ goto Exit;
-+ }
-+
-+ // determine whether index falls within capture or render devices
-+ if ( device >= renderDeviceCount ) {
-+ if ( mode != INPUT ) {
-+ errorType = RtAudioError::INVALID_USE;
-+ errorText_ = "RtApiWasapi::probeDeviceOpen: Capture device selected as output device.";
-+ goto Exit;
-+ }
-+
-+ // retrieve captureAudioClient from devicePtr
-+ IAudioClient*& captureAudioClient = ( ( WasapiHandle* ) stream_.apiHandle )->captureAudioClient;
-+
-+ hr = captureDevices->Item( device - renderDeviceCount, &devicePtr );
-+ if ( FAILED( hr ) ) {
-+ errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve capture device handle.";
-+ goto Exit;
-+ }
-+
-+ hr = devicePtr->Activate( __uuidof( IAudioClient ), CLSCTX_ALL,
-+ NULL, ( void** ) &captureAudioClient );
-+ if ( FAILED( hr ) ) {
-+ errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve device audio client.";
-+ goto Exit;
-+ }
-+
-+ hr = captureAudioClient->GetMixFormat( &deviceFormat );
-+ if ( FAILED( hr ) ) {
-+ errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve device mix format.";
-+ goto Exit;
-+ }
-+
-+ stream_.nDeviceChannels[mode] = deviceFormat->nChannels;
-+ captureAudioClient->GetStreamLatency( ( long long* ) &stream_.latency[mode] );
-+ }
-+ else {
-+ if ( mode != OUTPUT ) {
-+ errorType = RtAudioError::INVALID_USE;
-+ errorText_ = "RtApiWasapi::probeDeviceOpen: Render device selected as input device.";
-+ goto Exit;
-+ }
-+
-+ // retrieve renderAudioClient from devicePtr
-+ IAudioClient*& renderAudioClient = ( ( WasapiHandle* ) stream_.apiHandle )->renderAudioClient;
-+
-+ hr = renderDevices->Item( device, &devicePtr );
-+ if ( FAILED( hr ) ) {
-+ errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve render device handle.";
-+ goto Exit;
-+ }
-+
-+ hr = devicePtr->Activate( __uuidof( IAudioClient ), CLSCTX_ALL,
-+ NULL, ( void** ) &renderAudioClient );
-+ if ( FAILED( hr ) ) {
-+ errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve device audio client.";
-+ goto Exit;
-+ }
-+
-+ hr = renderAudioClient->GetMixFormat( &deviceFormat );
-+ if ( FAILED( hr ) ) {
-+ errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve device mix format.";
-+ goto Exit;
-+ }
-+
-+ stream_.nDeviceChannels[mode] = deviceFormat->nChannels;
-+ renderAudioClient->GetStreamLatency( ( long long* ) &stream_.latency[mode] );
-+ }
-+
-+ // fill stream data
-+ if ( ( stream_.mode == OUTPUT && mode == INPUT ) ||
-+ ( stream_.mode == INPUT && mode == OUTPUT ) ) {
-+ stream_.mode = DUPLEX;
-+ }
-+ else {
-+ stream_.mode = mode;
-+ }
-+
-+ stream_.device[mode] = device;
-+ stream_.doByteSwap[mode] = false;
-+ stream_.sampleRate = sampleRate;
-+ stream_.bufferSize = *bufferSize;
-+ stream_.nBuffers = 1;
-+ stream_.nUserChannels[mode] = channels;
-+ stream_.channelOffset[mode] = firstChannel;
-+ stream_.userFormat = format;
-+ stream_.deviceFormat[mode] = getDeviceInfo( device ).nativeFormats;
-+
-+ if ( options && options->flags & RTAUDIO_NONINTERLEAVED )
-+ stream_.userInterleaved = false;
-+ else
-+ stream_.userInterleaved = true;
-+ stream_.deviceInterleaved[mode] = true;
-+
-+ // Set flags for buffer conversion.
-+ stream_.doConvertBuffer[mode] = false;
-+ if ( stream_.userFormat != stream_.deviceFormat[mode] ||
-+ stream_.nUserChannels != stream_.nDeviceChannels )
-+ stream_.doConvertBuffer[mode] = true;
-+ else if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] &&
-+ stream_.nUserChannels[mode] > 1 )
-+ stream_.doConvertBuffer[mode] = true;
-+
-+ if ( stream_.doConvertBuffer[mode] )
-+ setConvertInfo( mode, 0 );
-+
-+ // Allocate necessary internal buffers
-+ bufferBytes = stream_.nUserChannels[mode] * stream_.bufferSize * formatBytes( stream_.userFormat );
-+
-+ stream_.userBuffer[mode] = ( char* ) calloc( bufferBytes, 1 );
-+ if ( !stream_.userBuffer[mode] ) {
-+ errorType = RtAudioError::MEMORY_ERROR;
-+ errorText_ = "RtApiWasapi::probeDeviceOpen: Error allocating user buffer memory.";
-+ goto Exit;
-+ }
-+
-+ if ( options && options->flags & RTAUDIO_SCHEDULE_REALTIME )
-+ stream_.callbackInfo.priority = 15;
-+ else
-+ stream_.callbackInfo.priority = 0;
-+
-+ ///! TODO: RTAUDIO_MINIMIZE_LATENCY // Provide stream buffers directly to callback
-+ ///! TODO: RTAUDIO_HOG_DEVICE // Exclusive mode
-+
-+ methodResult = SUCCESS;
-+
-+Exit:
-+ //clean up
-+ SAFE_RELEASE( captureDevices );
-+ SAFE_RELEASE( renderDevices );
-+ SAFE_RELEASE( devicePtr );
-+ CoTaskMemFree( deviceFormat );
-+
-+ // if method failed, close the stream
-+ if ( methodResult == FAILURE )
-+ closeStream();
-+
-+ if ( !errorText_.empty() )
-+ error( errorType );
-+ return methodResult;
-+}
-+
-+//=============================================================================
-+
-+DWORD WINAPI RtApiWasapi::runWasapiThread( void* wasapiPtr )
-+{
-+ if ( wasapiPtr )
-+ ( ( RtApiWasapi* ) wasapiPtr )->wasapiThread();
-+
-+ return 0;
-+}
-+
-+DWORD WINAPI RtApiWasapi::stopWasapiThread( void* wasapiPtr )
-+{
-+ if ( wasapiPtr )
-+ ( ( RtApiWasapi* ) wasapiPtr )->stopStream();
-+
-+ return 0;
-+}
-+
-+DWORD WINAPI RtApiWasapi::abortWasapiThread( void* wasapiPtr )
-+{
-+ if ( wasapiPtr )
-+ ( ( RtApiWasapi* ) wasapiPtr )->abortStream();
-+
-+ return 0;
-+}
-+
-+//-----------------------------------------------------------------------------
-+
-+void RtApiWasapi::wasapiThread()
-+{
-+ // as this is a new thread, we must CoInitialize it
-+ CoInitialize( NULL );
-+
-+ HRESULT hr;
-+
-+ IAudioClient* captureAudioClient = ( ( WasapiHandle* ) stream_.apiHandle )->captureAudioClient;
-+ IAudioClient* renderAudioClient = ( ( WasapiHandle* ) stream_.apiHandle )->renderAudioClient;
-+ IAudioCaptureClient* captureClient = ( ( WasapiHandle* ) stream_.apiHandle )->captureClient;
-+ IAudioRenderClient* renderClient = ( ( WasapiHandle* ) stream_.apiHandle )->renderClient;
-+ HANDLE captureEvent = ( ( WasapiHandle* ) stream_.apiHandle )->captureEvent;
-+ HANDLE renderEvent = ( ( WasapiHandle* ) stream_.apiHandle )->renderEvent;
-+
-+ WAVEFORMATEX* captureFormat = NULL;
-+ WAVEFORMATEX* renderFormat = NULL;
-+ float captureSrRatio = 0.0f;
-+ float renderSrRatio = 0.0f;
-+ WasapiBuffer captureBuffer;
-+ WasapiBuffer renderBuffer;
-+
-+ // declare local stream variables
-+ RtAudioCallback callback = ( RtAudioCallback ) stream_.callbackInfo.callback;
-+ BYTE* streamBuffer = NULL;
-+ unsigned long captureFlags = 0;
-+ unsigned int bufferFrameCount = 0;
-+ unsigned int numFramesPadding = 0;
-+ unsigned int convBufferSize = 0;
-+ bool callbackPushed = false;
-+ bool callbackPulled = false;
-+ bool callbackStopped = false;
-+ int callbackResult = 0;
-+
-+ // convBuffer is used to store converted buffers between WASAPI and the user
-+ char* convBuffer = NULL;
-+ unsigned int convBuffSize = 0;
-+ unsigned int deviceBuffSize = 0;
-+
-+ errorText_.clear();
-+ RtAudioError::Type errorType = RtAudioError::DRIVER_ERROR;
-+
-+ // Attempt to assign "Pro Audio" characteristic to thread
-+ HMODULE AvrtDll = LoadLibrary( (LPCTSTR) "AVRT.dll" );
-+ if ( AvrtDll ) {
-+ DWORD taskIndex = 0;
-+ TAvSetMmThreadCharacteristicsPtr AvSetMmThreadCharacteristicsPtr = ( TAvSetMmThreadCharacteristicsPtr ) GetProcAddress( AvrtDll, "AvSetMmThreadCharacteristicsW" );
-+ AvSetMmThreadCharacteristicsPtr( L"Pro Audio", &taskIndex );
-+ FreeLibrary( AvrtDll );
-+ }
-+
-+ // start capture stream if applicable
-+ if ( captureAudioClient ) {
-+ hr = captureAudioClient->GetMixFormat( &captureFormat );
-+ if ( FAILED( hr ) ) {
-+ errorText_ = "RtApiWasapi::wasapiThread: Unable to retrieve device mix format.";
-+ goto Exit;
-+ }
-+
-+ captureSrRatio = ( ( float ) captureFormat->nSamplesPerSec / stream_.sampleRate );
-+
-+ // initialize capture stream according to desire buffer size
-+ float desiredBufferSize = stream_.bufferSize * captureSrRatio;
-+ REFERENCE_TIME desiredBufferPeriod = ( REFERENCE_TIME ) ( ( float ) desiredBufferSize * 10000000 / captureFormat->nSamplesPerSec );
-+
-+ if ( !captureClient ) {
-+ hr = captureAudioClient->Initialize( AUDCLNT_SHAREMODE_SHARED,
-+ AUDCLNT_STREAMFLAGS_EVENTCALLBACK,
-+ desiredBufferPeriod,
-+ desiredBufferPeriod,
-+ captureFormat,
-+ NULL );
-+ if ( FAILED( hr ) ) {
-+ errorText_ = "RtApiWasapi::wasapiThread: Unable to initialize capture audio client.";
-+ goto Exit;
-+ }
-+
-+ hr = captureAudioClient->GetService( __uuidof( IAudioCaptureClient ),
-+ ( void** ) &captureClient );
-+ if ( FAILED( hr ) ) {
-+ errorText_ = "RtApiWasapi::wasapiThread: Unable to retrieve capture client handle.";
-+ goto Exit;
-+ }
-+
-+ // configure captureEvent to trigger on every available capture buffer
-+ captureEvent = CreateEvent( NULL, FALSE, FALSE, NULL );
-+ if ( !captureEvent ) {
-+ errorType = RtAudioError::SYSTEM_ERROR;
-+ errorText_ = "RtApiWasapi::wasapiThread: Unable to create capture event.";
-+ goto Exit;
-+ }
-+
-+ hr = captureAudioClient->SetEventHandle( captureEvent );
-+ if ( FAILED( hr ) ) {
-+ errorText_ = "RtApiWasapi::wasapiThread: Unable to set capture event handle.";
-+ goto Exit;
-+ }
-+
-+ ( ( WasapiHandle* ) stream_.apiHandle )->captureClient = captureClient;
-+ ( ( WasapiHandle* ) stream_.apiHandle )->captureEvent = captureEvent;
-+ }
-+
-+ unsigned int inBufferSize = 0;
-+ hr = captureAudioClient->GetBufferSize( &inBufferSize );
-+ if ( FAILED( hr ) ) {
-+ errorText_ = "RtApiWasapi::wasapiThread: Unable to get capture buffer size.";
-+ goto Exit;
-+ }
-+
-+ // scale outBufferSize according to stream->user sample rate ratio
-+ unsigned int outBufferSize = ( unsigned int ) ( stream_.bufferSize * captureSrRatio ) * stream_.nDeviceChannels[INPUT];
-+ inBufferSize *= stream_.nDeviceChannels[INPUT];
-+
-+ // set captureBuffer size
-+ captureBuffer.setBufferSize( inBufferSize + outBufferSize, formatBytes( stream_.deviceFormat[INPUT] ) );
-+
-+ // reset the capture stream
-+ hr = captureAudioClient->Reset();
-+ if ( FAILED( hr ) ) {
-+ errorText_ = "RtApiWasapi::wasapiThread: Unable to reset capture stream.";
-+ goto Exit;
-+ }
-+
-+ // start the capture stream
-+ hr = captureAudioClient->Start();
-+ if ( FAILED( hr ) ) {
-+ errorText_ = "RtApiWasapi::wasapiThread: Unable to start capture stream.";
-+ goto Exit;
-+ }
-+ }
-+
-+ // start render stream if applicable
-+ if ( renderAudioClient ) {
-+ hr = renderAudioClient->GetMixFormat( &renderFormat );
-+ if ( FAILED( hr ) ) {
-+ errorText_ = "RtApiWasapi::wasapiThread: Unable to retrieve device mix format.";
-+ goto Exit;
-+ }
-+
-+ renderSrRatio = ( ( float ) renderFormat->nSamplesPerSec / stream_.sampleRate );
-+
-+ // initialize render stream according to desire buffer size
-+ float desiredBufferSize = stream_.bufferSize * renderSrRatio;
-+ REFERENCE_TIME desiredBufferPeriod = ( REFERENCE_TIME ) ( ( float ) desiredBufferSize * 10000000 / renderFormat->nSamplesPerSec );
-+
-+ if ( !renderClient ) {
-+ hr = renderAudioClient->Initialize( AUDCLNT_SHAREMODE_SHARED,
-+ AUDCLNT_STREAMFLAGS_EVENTCALLBACK,
-+ desiredBufferPeriod,
-+ desiredBufferPeriod,
-+ renderFormat,
-+ NULL );
-+ if ( FAILED( hr ) ) {
-+ errorText_ = "RtApiWasapi::wasapiThread: Unable to initialize render audio client.";
-+ goto Exit;
-+ }
-+
-+ hr = renderAudioClient->GetService( __uuidof( IAudioRenderClient ),
-+ ( void** ) &renderClient );
-+ if ( FAILED( hr ) ) {
-+ errorText_ = "RtApiWasapi::wasapiThread: Unable to retrieve render client handle.";
-+ goto Exit;
-+ }
-+
-+ // configure renderEvent to trigger on every available render buffer
-+ renderEvent = CreateEvent( NULL, FALSE, FALSE, NULL );
-+ if ( !renderEvent ) {
-+ errorType = RtAudioError::SYSTEM_ERROR;
-+ errorText_ = "RtApiWasapi::wasapiThread: Unable to create render event.";
-+ goto Exit;
-+ }
-+
-+ hr = renderAudioClient->SetEventHandle( renderEvent );
-+ if ( FAILED( hr ) ) {
-+ errorText_ = "RtApiWasapi::wasapiThread: Unable to set render event handle.";
-+ goto Exit;
-+ }
-+
-+ ( ( WasapiHandle* ) stream_.apiHandle )->renderClient = renderClient;
-+ ( ( WasapiHandle* ) stream_.apiHandle )->renderEvent = renderEvent;
-+ }
-+
-+ unsigned int outBufferSize = 0;
-+ hr = renderAudioClient->GetBufferSize( &outBufferSize );
-+ if ( FAILED( hr ) ) {
-+ errorText_ = "RtApiWasapi::wasapiThread: Unable to get render buffer size.";
-+ goto Exit;
-+ }
-+
-+ // scale inBufferSize according to user->stream sample rate ratio
-+ unsigned int inBufferSize = ( unsigned int ) ( stream_.bufferSize * renderSrRatio ) * stream_.nDeviceChannels[OUTPUT];
-+ outBufferSize *= stream_.nDeviceChannels[OUTPUT];
-+
-+ // set renderBuffer size
-+ renderBuffer.setBufferSize( inBufferSize + outBufferSize, formatBytes( stream_.deviceFormat[OUTPUT] ) );
-+
-+ // reset the render stream
-+ hr = renderAudioClient->Reset();
-+ if ( FAILED( hr ) ) {
-+ errorText_ = "RtApiWasapi::wasapiThread: Unable to reset render stream.";
-+ goto Exit;
-+ }
-+
-+ // start the render stream
-+ hr = renderAudioClient->Start();
-+ if ( FAILED( hr ) ) {
-+ errorText_ = "RtApiWasapi::wasapiThread: Unable to start render stream.";
-+ goto Exit;
-+ }
-+ }
-+
-+ if ( stream_.mode == INPUT ) {
-+ convBuffSize = ( size_t ) ( stream_.bufferSize * captureSrRatio ) * stream_.nDeviceChannels[INPUT] * formatBytes( stream_.deviceFormat[INPUT] );
-+ deviceBuffSize = stream_.bufferSize * stream_.nDeviceChannels[INPUT] * formatBytes( stream_.deviceFormat[INPUT] );
-+ }
-+ else if ( stream_.mode == OUTPUT ) {
-+ convBuffSize = ( size_t ) ( stream_.bufferSize * renderSrRatio ) * stream_.nDeviceChannels[OUTPUT] * formatBytes( stream_.deviceFormat[OUTPUT] );
-+ deviceBuffSize = stream_.bufferSize * stream_.nDeviceChannels[OUTPUT] * formatBytes( stream_.deviceFormat[OUTPUT] );
-+ }
-+ else if ( stream_.mode == DUPLEX ) {
-+ convBuffSize = std::max( ( size_t ) ( stream_.bufferSize * captureSrRatio ) * stream_.nDeviceChannels[INPUT] * formatBytes( stream_.deviceFormat[INPUT] ),
-+ ( size_t ) ( stream_.bufferSize * renderSrRatio ) * stream_.nDeviceChannels[OUTPUT] * formatBytes( stream_.deviceFormat[OUTPUT] ) );
-+ deviceBuffSize = std::max( stream_.bufferSize * stream_.nDeviceChannels[INPUT] * formatBytes( stream_.deviceFormat[INPUT] ),
-+ stream_.bufferSize * stream_.nDeviceChannels[OUTPUT] * formatBytes( stream_.deviceFormat[OUTPUT] ) );
-+ }
-+
-+ convBuffer = ( char* ) malloc( convBuffSize );
-+ stream_.deviceBuffer = ( char* ) malloc( deviceBuffSize );
-+ if ( !convBuffer || !stream_.deviceBuffer ) {
-+ errorType = RtAudioError::MEMORY_ERROR;
-+ errorText_ = "RtApiWasapi::wasapiThread: Error allocating device buffer memory.";
-+ goto Exit;
-+ }
-+
-+ // stream process loop
-+ while ( stream_.state != STREAM_STOPPING ) {
-+ if ( !callbackPulled ) {
-+ // Callback Input
-+ // ==============
-+ // 1. Pull callback buffer from inputBuffer
-+ // 2. If 1. was successful: Convert callback buffer to user sample rate and channel count
-+ // Convert callback buffer to user format
-+
-+ if ( captureAudioClient ) {
-+ // Pull callback buffer from inputBuffer
-+ callbackPulled = captureBuffer.pullBuffer( convBuffer,
-+ ( unsigned int ) ( stream_.bufferSize * captureSrRatio ) * stream_.nDeviceChannels[INPUT],
-+ stream_.deviceFormat[INPUT] );
-+
-+ if ( callbackPulled ) {
-+ // Convert callback buffer to user sample rate
-+ convertBufferWasapi( stream_.deviceBuffer,
-+ convBuffer,
-+ stream_.nDeviceChannels[INPUT],
-+ captureFormat->nSamplesPerSec,
-+ stream_.sampleRate,
-+ ( unsigned int ) ( stream_.bufferSize * captureSrRatio ),
-+ convBufferSize,
-+ stream_.deviceFormat[INPUT] );
-+
-+ if ( stream_.doConvertBuffer[INPUT] ) {
-+ // Convert callback buffer to user format
-+ convertBuffer( stream_.userBuffer[INPUT],
-+ stream_.deviceBuffer,
-+ stream_.convertInfo[INPUT] );
-+ }
-+ else {
-+ // no further conversion, simple copy deviceBuffer to userBuffer
-+ memcpy( stream_.userBuffer[INPUT],
-+ stream_.deviceBuffer,
-+ stream_.bufferSize * stream_.nUserChannels[INPUT] * formatBytes( stream_.userFormat ) );
-+ }
-+ }
-+ }
-+ else {
-+ // if there is no capture stream, set callbackPulled flag
-+ callbackPulled = true;
-+ }
-+
-+ // Execute Callback
-+ // ================
-+ // 1. Execute user callback method
-+ // 2. Handle return value from callback
-+
-+ // if callback has not requested the stream to stop
-+ if ( callbackPulled && !callbackStopped ) {
-+ // Execute user callback method
-+ callbackResult = callback( stream_.userBuffer[OUTPUT],
-+ stream_.userBuffer[INPUT],
-+ stream_.bufferSize,
-+ getStreamTime(),
-+ captureFlags & AUDCLNT_BUFFERFLAGS_DATA_DISCONTINUITY ? RTAUDIO_INPUT_OVERFLOW : 0,
-+ stream_.callbackInfo.userData );
-+
-+ // Handle return value from callback
-+ if ( callbackResult == 1 ) {
-+ // instantiate a thread to stop this thread
-+ HANDLE threadHandle = CreateThread( NULL, 0, stopWasapiThread, this, 0, NULL );
-+ if ( !threadHandle ) {
-+ errorType = RtAudioError::THREAD_ERROR;
-+ errorText_ = "RtApiWasapi::wasapiThread: Unable to instantiate stream stop thread.";
-+ goto Exit;
-+ }
-+ else if ( !CloseHandle( threadHandle ) ) {
-+ errorType = RtAudioError::THREAD_ERROR;
-+ errorText_ = "RtApiWasapi::wasapiThread: Unable to close stream stop thread handle.";
-+ goto Exit;
-+ }
-+
-+ callbackStopped = true;
-+ }
-+ else if ( callbackResult == 2 ) {
-+ // instantiate a thread to stop this thread
-+ HANDLE threadHandle = CreateThread( NULL, 0, abortWasapiThread, this, 0, NULL );
-+ if ( !threadHandle ) {
-+ errorType = RtAudioError::THREAD_ERROR;
-+ errorText_ = "RtApiWasapi::wasapiThread: Unable to instantiate stream abort thread.";
-+ goto Exit;
-+ }
-+ else if ( !CloseHandle( threadHandle ) ) {
-+ errorType = RtAudioError::THREAD_ERROR;
-+ errorText_ = "RtApiWasapi::wasapiThread: Unable to close stream abort thread handle.";
-+ goto Exit;
-+ }
-+
-+ callbackStopped = true;
-+ }
-+ }
-+ }
-+
-+ // Callback Output
-+ // ===============
-+ // 1. Convert callback buffer to stream format
-+ // 2. Convert callback buffer to stream sample rate and channel count
-+ // 3. Push callback buffer into outputBuffer
-+
-+ if ( renderAudioClient && callbackPulled ) {
-+ if ( stream_.doConvertBuffer[OUTPUT] ) {
-+ // Convert callback buffer to stream format
-+ convertBuffer( stream_.deviceBuffer,
-+ stream_.userBuffer[OUTPUT],
-+ stream_.convertInfo[OUTPUT] );
-+
-+ }
-+
-+ // Convert callback buffer to stream sample rate
-+ convertBufferWasapi( convBuffer,
-+ stream_.deviceBuffer,
-+ stream_.nDeviceChannels[OUTPUT],
-+ stream_.sampleRate,
-+ renderFormat->nSamplesPerSec,
-+ stream_.bufferSize,
-+ convBufferSize,
-+ stream_.deviceFormat[OUTPUT] );
-+
-+ // Push callback buffer into outputBuffer
-+ callbackPushed = renderBuffer.pushBuffer( convBuffer,
-+ convBufferSize * stream_.nDeviceChannels[OUTPUT],
-+ stream_.deviceFormat[OUTPUT] );
-+ }
-+ else {
-+ // if there is no render stream, set callbackPushed flag
-+ callbackPushed = true;
-+ }
-+
-+ // Stream Capture
-+ // ==============
-+ // 1. Get capture buffer from stream
-+ // 2. Push capture buffer into inputBuffer
-+ // 3. If 2. was successful: Release capture buffer
-+
-+ if ( captureAudioClient ) {
-+ // if the callback input buffer was not pulled from captureBuffer, wait for next capture event
-+ if ( !callbackPulled ) {
-+ WaitForSingleObject( captureEvent, INFINITE );
-+ }
-+
-+ // Get capture buffer from stream
-+ hr = captureClient->GetBuffer( &streamBuffer,
-+ &bufferFrameCount,
-+ &captureFlags, NULL, NULL );
-+ if ( FAILED( hr ) ) {
-+ errorText_ = "RtApiWasapi::wasapiThread: Unable to retrieve capture buffer.";
-+ goto Exit;
-+ }
-+
-+ if ( bufferFrameCount != 0 ) {
-+ // Push capture buffer into inputBuffer
-+ if ( captureBuffer.pushBuffer( ( char* ) streamBuffer,
-+ bufferFrameCount * stream_.nDeviceChannels[INPUT],
-+ stream_.deviceFormat[INPUT] ) )
-+ {
-+ // Release capture buffer
-+ hr = captureClient->ReleaseBuffer( bufferFrameCount );
-+ if ( FAILED( hr ) ) {
-+ errorText_ = "RtApiWasapi::wasapiThread: Unable to release capture buffer.";
-+ goto Exit;
-+ }
-+ }
-+ else
-+ {
-+ // Inform WASAPI that capture was unsuccessful
-+ hr = captureClient->ReleaseBuffer( 0 );
-+ if ( FAILED( hr ) ) {
-+ errorText_ = "RtApiWasapi::wasapiThread: Unable to release capture buffer.";
-+ goto Exit;
-+ }
-+ }
-+ }
-+ else
-+ {
-+ // Inform WASAPI that capture was unsuccessful
-+ hr = captureClient->ReleaseBuffer( 0 );
-+ if ( FAILED( hr ) ) {
-+ errorText_ = "RtApiWasapi::wasapiThread: Unable to release capture buffer.";
-+ goto Exit;
-+ }
-+ }
-+ }
-+
-+ // Stream Render
-+ // =============
-+ // 1. Get render buffer from stream
-+ // 2. Pull next buffer from outputBuffer
-+ // 3. If 2. was successful: Fill render buffer with next buffer
-+ // Release render buffer
-+
-+ if ( renderAudioClient ) {
-+ // if the callback output buffer was not pushed to renderBuffer, wait for next render event
-+ if ( callbackPulled && !callbackPushed ) {
-+ WaitForSingleObject( renderEvent, INFINITE );
-+ }
-+
-+ // Get render buffer from stream
-+ hr = renderAudioClient->GetBufferSize( &bufferFrameCount );
-+ if ( FAILED( hr ) ) {
-+ errorText_ = "RtApiWasapi::wasapiThread: Unable to retrieve render buffer size.";
-+ goto Exit;
-+ }
-+
-+ hr = renderAudioClient->GetCurrentPadding( &numFramesPadding );
-+ if ( FAILED( hr ) ) {
-+ errorText_ = "RtApiWasapi::wasapiThread: Unable to retrieve render buffer padding.";
-+ goto Exit;
-+ }
-+
-+ bufferFrameCount -= numFramesPadding;
-+
-+ if ( bufferFrameCount != 0 ) {
-+ hr = renderClient->GetBuffer( bufferFrameCount, &streamBuffer );
-+ if ( FAILED( hr ) ) {
-+ errorText_ = "RtApiWasapi::wasapiThread: Unable to retrieve render buffer.";
-+ goto Exit;
-+ }
-+
-+ // Pull next buffer from outputBuffer
-+ // Fill render buffer with next buffer
-+ if ( renderBuffer.pullBuffer( ( char* ) streamBuffer,
-+ bufferFrameCount * stream_.nDeviceChannels[OUTPUT],
-+ stream_.deviceFormat[OUTPUT] ) )
-+ {
-+ // Release render buffer
-+ hr = renderClient->ReleaseBuffer( bufferFrameCount, 0 );
-+ if ( FAILED( hr ) ) {
-+ errorText_ = "RtApiWasapi::wasapiThread: Unable to release render buffer.";
-+ goto Exit;
-+ }
-+ }
-+ else
-+ {
-+ // Inform WASAPI that render was unsuccessful
-+ hr = renderClient->ReleaseBuffer( 0, 0 );
-+ if ( FAILED( hr ) ) {
-+ errorText_ = "RtApiWasapi::wasapiThread: Unable to release render buffer.";
-+ goto Exit;
-+ }
-+ }
-+ }
-+ else
-+ {
-+ // Inform WASAPI that render was unsuccessful
-+ hr = renderClient->ReleaseBuffer( 0, 0 );
-+ if ( FAILED( hr ) ) {
-+ errorText_ = "RtApiWasapi::wasapiThread: Unable to release render buffer.";
-+ goto Exit;
-+ }
-+ }
-+ }
-+
-+ // if the callback buffer was pushed renderBuffer reset callbackPulled flag
-+ if ( callbackPushed ) {
-+ callbackPulled = false;
-+ // tick stream time
-+ RtApi::tickStreamTime();
-+ }
-+
-+ }
-+
-+Exit:
-+ // clean up
-+ CoTaskMemFree( captureFormat );
-+ CoTaskMemFree( renderFormat );
-+
-+ free ( convBuffer );
-+
-+ CoUninitialize();
-+
-+ // update stream state
-+ stream_.state = STREAM_STOPPED;
-+
-+ if ( errorText_.empty() )
-+ return;
-+ else
-+ error( errorType );
-+}
-+
-+//******************** End of __WINDOWS_WASAPI__ *********************//
-+#endif
-+
-+
-+#if defined(__WINDOWS_DS__) // Windows DirectSound API
-+
-+// Modified by Robin Davies, October 2005
-+// - Improvements to DirectX pointer chasing.
-+// - Bug fix for non-power-of-two Asio granularity used by Edirol PCR-A30.
-+// - Auto-call CoInitialize for DSOUND and ASIO platforms.
-+// Various revisions for RtAudio 4.0 by Gary Scavone, April 2007
-+// Changed device query structure for RtAudio 4.0.7, January 2010
-+
-+#include <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 );
-+ }
-+
-+ result = snd_ctl_open( &chandle, "default", SND_CTL_NONBLOCK );
-+ if ( result == 0 ) {
-+ if ( nDevices == device ) {
-+ strcpy( name, "default" );
-+ goto foundDevice;
-+ }
-+ nDevices++;
-+ }
-+
-+ if ( nDevices == 0 ) {
-+ errorText_ = "RtApiAlsa::getDeviceInfo: no devices found!";
-+ error( RtAudioError::INVALID_USE );
-+ return info;
-+ }
-+
-+ if ( device >= nDevices ) {
-+ errorText_ = "RtApiAlsa::getDeviceInfo: device ID is invalid!";
-+ error( RtAudioError::INVALID_USE );
-+ return info;
-+ }
-+
-+ foundDevice:
-+
-+ // If a stream is already open, we cannot probe the stream devices.
-+ // Thus, use the saved results.
-+ if ( stream_.state != STREAM_CLOSED &&
-+ ( stream_.device[0] == device || stream_.device[1] == device ) ) {
-+ snd_ctl_close( chandle );
-+ if ( device >= devices_.size() ) {
-+ errorText_ = "RtApiAlsa::getDeviceInfo: device ID was not present before stream was opened.";
-+ error( RtAudioError::WARNING );
-+ return info;
-+ }
-+ return devices_[ device ];
-+ }
-+
-+ int openMode = SND_PCM_ASYNC;
-+ snd_pcm_stream_t stream;
-+ snd_pcm_info_t *pcminfo;
-+ snd_pcm_info_alloca( &pcminfo );
-+ snd_pcm_t *phandle;
-+ snd_pcm_hw_params_t *params;
-+ snd_pcm_hw_params_alloca( ¶ms );
-+
-+ // First try for playback unless default device (which has subdev -1)
-+ stream = SND_PCM_STREAM_PLAYBACK;
-+ snd_pcm_info_set_stream( pcminfo, stream );
-+ if ( subdevice != -1 ) {
-+ snd_pcm_info_set_device( pcminfo, subdevice );
-+ snd_pcm_info_set_subdevice( pcminfo, 0 );
-+
-+ result = snd_ctl_pcm_info( chandle, pcminfo );
-+ if ( result < 0 ) {
-+ // Device probably doesn't support playback.
-+ goto captureProbe;
-+ }
-+ }
-+
-+ result = snd_pcm_open( &phandle, name, stream, openMode | SND_PCM_NONBLOCK );
-+ if ( result < 0 ) {
-+ errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_open error for device (" << name << "), " << snd_strerror( result ) << ".";
-+ errorText_ = errorStream_.str();
-+ error( RtAudioError::WARNING );
-+ goto captureProbe;
-+ }
-+
-+ // The device is open ... fill the parameter structure.
-+ result = snd_pcm_hw_params_any( phandle, params );
-+ if ( result < 0 ) {
-+ snd_pcm_close( phandle );
-+ errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_hw_params error for device (" << name << "), " << snd_strerror( result ) << ".";
-+ errorText_ = errorStream_.str();
-+ error( RtAudioError::WARNING );
-+ goto captureProbe;
-+ }
-+
-+ // Get output channel information.
-+ unsigned int value;
-+ result = snd_pcm_hw_params_get_channels_max( params, &value );
-+ if ( result < 0 ) {
-+ snd_pcm_close( phandle );
-+ errorStream_ << "RtApiAlsa::getDeviceInfo: error getting device (" << name << ") output channels, " << snd_strerror( result ) << ".";
-+ errorText_ = errorStream_.str();
-+ error( RtAudioError::WARNING );
-+ goto captureProbe;
-+ }
-+ info.outputChannels = value;
-+ snd_pcm_close( phandle );
-+
-+ captureProbe:
-+ stream = SND_PCM_STREAM_CAPTURE;
-+ snd_pcm_info_set_stream( pcminfo, stream );
-+
-+ // Now try for capture unless default device (with subdev = -1)
-+ if ( subdevice != -1 ) {
-+ result = snd_ctl_pcm_info( chandle, pcminfo );
-+ snd_ctl_close( chandle );
-+ if ( result < 0 ) {
-+ // Device probably doesn't support capture.
-+ if ( info.outputChannels == 0 ) return info;
-+ goto probeParameters;
-+ }
-+ }
-+ else
-+ snd_ctl_close( chandle );
-+
-+ result = snd_pcm_open( &phandle, name, stream, openMode | SND_PCM_NONBLOCK);
-+ if ( result < 0 ) {
-+ errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_open error for device (" << name << "), " << snd_strerror( result ) << ".";
-+ errorText_ = errorStream_.str();
-+ error( RtAudioError::WARNING );
-+ if ( info.outputChannels == 0 ) return info;
-+ goto probeParameters;
-+ }
-+
-+ // The device is open ... fill the parameter structure.
-+ result = snd_pcm_hw_params_any( phandle, params );
-+ if ( result < 0 ) {
-+ snd_pcm_close( phandle );
-+ errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_hw_params error for device (" << name << "), " << snd_strerror( result ) << ".";
-+ errorText_ = errorStream_.str();
-+ error( RtAudioError::WARNING );
-+ if ( info.outputChannels == 0 ) return info;
-+ goto probeParameters;
-+ }
-+
-+ result = snd_pcm_hw_params_get_channels_max( params, &value );
-+ if ( result < 0 ) {
-+ snd_pcm_close( phandle );
-+ errorStream_ << "RtApiAlsa::getDeviceInfo: error getting device (" << name << ") input channels, " << snd_strerror( result ) << ".";
-+ errorText_ = errorStream_.str();
-+ error( RtAudioError::WARNING );
-+ if ( info.outputChannels == 0 ) return info;
-+ goto probeParameters;
-+ }
-+ info.inputChannels = value;
-+ snd_pcm_close( phandle );
-+
-+ // If device opens for both playback and capture, we determine the channels.
-+ if ( info.outputChannels > 0 && info.inputChannels > 0 )
-+ info.duplexChannels = (info.outputChannels > info.inputChannels) ? info.inputChannels : info.outputChannels;
-+
-+ // ALSA doesn't provide default devices so we'll use the first available one.
-+ if ( device == 0 && info.outputChannels > 0 )
-+ info.isDefaultOutput = true;
-+ if ( device == 0 && info.inputChannels > 0 )
-+ info.isDefaultInput = true;
-+
-+ probeParameters:
-+ // At this point, we just need to figure out the supported data
-+ // formats and sample rates. We'll proceed by opening the device in
-+ // the direction with the maximum number of channels, or playback if
-+ // they are equal. This might limit our sample rate options, but so
-+ // be it.
-+
-+ if ( info.outputChannels >= info.inputChannels )
-+ stream = SND_PCM_STREAM_PLAYBACK;
-+ else
-+ stream = SND_PCM_STREAM_CAPTURE;
-+ snd_pcm_info_set_stream( pcminfo, stream );
-+
-+ result = snd_pcm_open( &phandle, name, stream, openMode | SND_PCM_NONBLOCK);
-+ if ( result < 0 ) {
-+ errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_open error for device (" << name << "), " << snd_strerror( result ) << ".";
-+ errorText_ = errorStream_.str();
-+ error( RtAudioError::WARNING );
-+ return info;
-+ }
-+
-+ // The device is open ... fill the parameter structure.
-+ result = snd_pcm_hw_params_any( phandle, params );
-+ if ( result < 0 ) {
-+ snd_pcm_close( phandle );
-+ errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_hw_params error for device (" << name << "), " << snd_strerror( result ) << ".";
-+ errorText_ = errorStream_.str();
-+ error( RtAudioError::WARNING );
-+ return info;
-+ }
-+
-+ // Test our discrete set of sample rate values.
-+ info.sampleRates.clear();
-+ for ( unsigned int i=0; i<MAX_SAMPLE_RATES; i++ ) {
-+ if ( snd_pcm_hw_params_test_rate( phandle, params, SAMPLE_RATES[i], 0 ) == 0 ) {
-+ info.sampleRates.push_back( SAMPLE_RATES[i] );
-+
-+ if ( !info.preferredSampleRate || ( SAMPLE_RATES[i] <= 48000 && SAMPLE_RATES[i] > info.preferredSampleRate ) )
-+ info.preferredSampleRate = SAMPLE_RATES[i];
-+ }
-+ }
-+ if ( info.sampleRates.size() == 0 ) {
-+ snd_pcm_close( phandle );
-+ errorStream_ << "RtApiAlsa::getDeviceInfo: no supported sample rates found for device (" << name << ").";
-+ errorText_ = errorStream_.str();
-+ error( RtAudioError::WARNING );
-+ return info;
-+ }
-+
-+ // Probe the supported data formats ... we don't care about endian-ness just yet
-+ snd_pcm_format_t format;
-+ info.nativeFormats = 0;
-+ format = SND_PCM_FORMAT_S8;
-+ if ( snd_pcm_hw_params_test_format( phandle, params, format ) == 0 )
-+ info.nativeFormats |= RTAUDIO_SINT8;
-+ format = SND_PCM_FORMAT_S16;
-+ if ( snd_pcm_hw_params_test_format( phandle, params, format ) == 0 )
-+ info.nativeFormats |= RTAUDIO_SINT16;
-+ format = SND_PCM_FORMAT_S24;
-+ if ( snd_pcm_hw_params_test_format( phandle, params, format ) == 0 )
-+ info.nativeFormats |= RTAUDIO_SINT24;
-+ format = SND_PCM_FORMAT_S32;
-+ if ( snd_pcm_hw_params_test_format( phandle, params, format ) == 0 )
-+ info.nativeFormats |= RTAUDIO_SINT32;
-+ format = SND_PCM_FORMAT_FLOAT;
-+ if ( snd_pcm_hw_params_test_format( phandle, params, format ) == 0 )
-+ info.nativeFormats |= RTAUDIO_FLOAT32;
-+ format = SND_PCM_FORMAT_FLOAT64;
-+ if ( snd_pcm_hw_params_test_format( phandle, params, format ) == 0 )
-+ info.nativeFormats |= RTAUDIO_FLOAT64;
-+
-+ // Check that we have at least one supported format
-+ if ( info.nativeFormats == 0 ) {
-+ snd_pcm_close( phandle );
-+ errorStream_ << "RtApiAlsa::getDeviceInfo: pcm device (" << name << ") data format not supported by RtAudio.";
-+ errorText_ = errorStream_.str();
-+ error( RtAudioError::WARNING );
-+ return info;
-+ }
-+
-+ // Get the device name
-+ char *cardname;
-+ result = snd_card_get_name( card, &cardname );
-+ if ( result >= 0 ) {
-+ sprintf( name, "hw:%s,%d", cardname, subdevice );
-+ free( cardname );
-+ }
-+ info.name = name;
-+
-+ // That's all ... close the device and return
-+ snd_pcm_close( phandle );
-+ info.probed = true;
-+ return info;
-+}
-+
-+void RtApiAlsa :: saveDeviceInfo( void )
-+{
-+ devices_.clear();
-+
-+ unsigned int nDevices = getDeviceCount();
-+ devices_.resize( nDevices );
-+ for ( unsigned int i=0; i<nDevices; i++ )
-+ devices_[i] = getDeviceInfo( i );
-+}
-+
-+bool RtApiAlsa :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,
-+ unsigned int firstChannel, unsigned int sampleRate,
-+ RtAudioFormat format, unsigned int *bufferSize,
-+ RtAudio::StreamOptions *options )
-+
-+{
-+#if defined(__RTAUDIO_DEBUG__)
-+ snd_output_t *out;
-+ snd_output_stdio_attach(&out, stderr, 0);
-+#endif
-+
-+ // I'm not using the "plug" interface ... too much inconsistent behavior.
-+
-+ unsigned nDevices = 0;
-+ int result, subdevice, card;
-+ char name[64];
-+ snd_ctl_t *chandle;
-+
-+ if ( options && options->flags & RTAUDIO_ALSA_USE_DEFAULT )
-+ snprintf(name, sizeof(name), "%s", "default");
-+ else {
-+ // Count cards and devices
-+ card = -1;
-+ snd_card_next( &card );
-+ while ( card >= 0 ) {
-+ sprintf( name, "hw:%d", card );
-+ result = snd_ctl_open( &chandle, name, SND_CTL_NONBLOCK );
-+ if ( result < 0 ) {
-+ errorStream_ << "RtApiAlsa::probeDeviceOpen: control open, card = " << card << ", " << snd_strerror( result ) << ".";
-+ errorText_ = errorStream_.str();
-+ return FAILURE;
-+ }
-+ subdevice = -1;
-+ while( 1 ) {
-+ result = snd_ctl_pcm_next_device( chandle, &subdevice );
-+ if ( result < 0 ) break;
-+ if ( subdevice < 0 ) break;
-+ if ( nDevices == device ) {
-+ sprintf( name, "hw:%d,%d", card, subdevice );
-+ snd_ctl_close( chandle );
-+ goto foundDevice;
-+ }
-+ nDevices++;
-+ }
-+ snd_ctl_close( chandle );
-+ snd_card_next( &card );
-+ }
-+
-+ result = snd_ctl_open( &chandle, "default", SND_CTL_NONBLOCK );
-+ if ( result == 0 ) {
-+ if ( nDevices == device ) {
-+ strcpy( name, "default" );
-+ goto foundDevice;
-+ }
-+ nDevices++;
-+ }
-+
-+ if ( nDevices == 0 ) {
-+ // This should not happen because a check is made before this function is called.
-+ errorText_ = "RtApiAlsa::probeDeviceOpen: no devices found!";
-+ return FAILURE;
-+ }
-+
-+ if ( device >= nDevices ) {
-+ // This should not happen because a check is made before this function is called.
-+ errorText_ = "RtApiAlsa::probeDeviceOpen: device ID is invalid!";
-+ return FAILURE;
-+ }
-+ }
-+
-+ foundDevice:
-+
-+ // The getDeviceInfo() function will not work for a device that is
-+ // already open. Thus, we'll probe the system before opening a
-+ // stream and save the results for use by getDeviceInfo().
-+ if ( mode == OUTPUT || ( mode == INPUT && stream_.mode != OUTPUT ) ) // only do once
-+ this->saveDeviceInfo();
-+
-+ snd_pcm_stream_t stream;
-+ if ( mode == OUTPUT )
-+ stream = SND_PCM_STREAM_PLAYBACK;
-+ else
-+ stream = SND_PCM_STREAM_CAPTURE;
-+
-+ snd_pcm_t *phandle;
-+ int openMode = SND_PCM_ASYNC;
-+ result = snd_pcm_open( &phandle, name, stream, openMode );
-+ if ( result < 0 ) {
-+ if ( mode == OUTPUT )
-+ errorStream_ << "RtApiAlsa::probeDeviceOpen: pcm device (" << name << ") won't open for output.";
-+ else
-+ errorStream_ << "RtApiAlsa::probeDeviceOpen: pcm device (" << name << ") won't open for input.";
-+ errorText_ = errorStream_.str();
-+ return FAILURE;
-+ }
-+
-+ // Fill the parameter structure.
-+ snd_pcm_hw_params_t *hw_params;
-+ snd_pcm_hw_params_alloca( &hw_params );
-+ result = snd_pcm_hw_params_any( phandle, hw_params );
-+ if ( result < 0 ) {
-+ snd_pcm_close( phandle );
-+ errorStream_ << "RtApiAlsa::probeDeviceOpen: error getting pcm device (" << name << ") parameters, " << snd_strerror( result ) << ".";
-+ errorText_ = errorStream_.str();
-+ return FAILURE;
-+ }
-+
-+#if defined(__RTAUDIO_DEBUG__)
-+ fprintf( stderr, "\nRtApiAlsa: dump hardware params just after device open:\n\n" );
-+ snd_pcm_hw_params_dump( hw_params, out );
-+#endif
-+
-+ // Set access ... check user preference.
-+ if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) {
-+ stream_.userInterleaved = false;
-+ result = snd_pcm_hw_params_set_access( phandle, hw_params, SND_PCM_ACCESS_RW_NONINTERLEAVED );
-+ if ( result < 0 ) {
-+ result = snd_pcm_hw_params_set_access( phandle, hw_params, SND_PCM_ACCESS_RW_INTERLEAVED );
-+ stream_.deviceInterleaved[mode] = true;
-+ }
-+ else
-+ stream_.deviceInterleaved[mode] = false;
-+ }
-+ else {
-+ stream_.userInterleaved = true;
-+ result = snd_pcm_hw_params_set_access( phandle, hw_params, SND_PCM_ACCESS_RW_INTERLEAVED );
-+ if ( result < 0 ) {
-+ result = snd_pcm_hw_params_set_access( phandle, hw_params, SND_PCM_ACCESS_RW_NONINTERLEAVED );
-+ stream_.deviceInterleaved[mode] = false;
-+ }
-+ else
-+ stream_.deviceInterleaved[mode] = true;
-+ }
-+
-+ if ( result < 0 ) {
-+ snd_pcm_close( phandle );
-+ errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting pcm device (" << name << ") access, " << snd_strerror( result ) << ".";
-+ errorText_ = errorStream_.str();
-+ return FAILURE;
-+ }
-+
-+ // Determine how to set the device format.
-+ stream_.userFormat = format;
-+ snd_pcm_format_t deviceFormat = SND_PCM_FORMAT_UNKNOWN;
-+
-+ if ( format == RTAUDIO_SINT8 )
-+ deviceFormat = SND_PCM_FORMAT_S8;
-+ else if ( format == RTAUDIO_SINT16 )
-+ deviceFormat = SND_PCM_FORMAT_S16;
-+ else if ( format == RTAUDIO_SINT24 )
-+ deviceFormat = SND_PCM_FORMAT_S24;
-+ else if ( format == RTAUDIO_SINT32 )
-+ deviceFormat = SND_PCM_FORMAT_S32;
-+ else if ( format == RTAUDIO_FLOAT32 )
-+ deviceFormat = SND_PCM_FORMAT_FLOAT;
-+ else if ( format == RTAUDIO_FLOAT64 )
-+ deviceFormat = SND_PCM_FORMAT_FLOAT64;
-+
-+ if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat) == 0) {
-+ stream_.deviceFormat[mode] = format;
-+ goto setFormat;
-+ }
-+
-+ // The user requested format is not natively supported by the device.
-+ deviceFormat = SND_PCM_FORMAT_FLOAT64;
-+ if ( snd_pcm_hw_params_test_format( phandle, hw_params, deviceFormat ) == 0 ) {
-+ stream_.deviceFormat[mode] = RTAUDIO_FLOAT64;
-+ goto setFormat;
-+ }
-+
-+ deviceFormat = SND_PCM_FORMAT_FLOAT;
-+ if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat ) == 0 ) {
-+ stream_.deviceFormat[mode] = RTAUDIO_FLOAT32;
-+ goto setFormat;
-+ }
-+
-+ deviceFormat = SND_PCM_FORMAT_S32;
-+ if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat ) == 0 ) {
-+ stream_.deviceFormat[mode] = RTAUDIO_SINT32;
-+ goto setFormat;
-+ }
-+
-+ deviceFormat = SND_PCM_FORMAT_S24;
-+ if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat ) == 0 ) {
-+ stream_.deviceFormat[mode] = RTAUDIO_SINT24;
-+ goto setFormat;
-+ }
-+
-+ deviceFormat = SND_PCM_FORMAT_S16;
-+ if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat ) == 0 ) {
-+ stream_.deviceFormat[mode] = RTAUDIO_SINT16;
-+ goto setFormat;
-+ }
-+
-+ deviceFormat = SND_PCM_FORMAT_S8;
-+ if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat ) == 0 ) {
-+ stream_.deviceFormat[mode] = RTAUDIO_SINT8;
-+ goto setFormat;
-+ }
-+
-+ // If we get here, no supported format was found.
-+ snd_pcm_close( phandle );
-+ errorStream_ << "RtApiAlsa::probeDeviceOpen: pcm device " << device << " data format not supported by RtAudio.";
-+ errorText_ = errorStream_.str();
-+ return FAILURE;
-+
-+ setFormat:
-+ result = snd_pcm_hw_params_set_format( phandle, hw_params, deviceFormat );
-+ if ( result < 0 ) {
-+ snd_pcm_close( phandle );
-+ errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting pcm device (" << name << ") data format, " << snd_strerror( result ) << ".";
-+ errorText_ = errorStream_.str();
-+ return FAILURE;
-+ }
-+
-+ // Determine whether byte-swaping is necessary.
-+ stream_.doByteSwap[mode] = false;
-+ if ( deviceFormat != SND_PCM_FORMAT_S8 ) {
-+ result = snd_pcm_format_cpu_endian( deviceFormat );
-+ if ( result == 0 )
-+ stream_.doByteSwap[mode] = true;
-+ else if (result < 0) {
-+ snd_pcm_close( phandle );
-+ errorStream_ << "RtApiAlsa::probeDeviceOpen: error getting pcm device (" << name << ") endian-ness, " << snd_strerror( result ) << ".";
-+ errorText_ = errorStream_.str();
-+ return FAILURE;
-+ }
-+ }
-+
-+ // Set the sample rate.
-+ result = snd_pcm_hw_params_set_rate_near( phandle, hw_params, (unsigned int*) &sampleRate, 0 );
-+ if ( result < 0 ) {
-+ snd_pcm_close( phandle );
-+ errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting sample rate on device (" << name << "), " << snd_strerror( result ) << ".";
-+ errorText_ = errorStream_.str();
-+ return FAILURE;
-+ }
-+
-+ // Determine the number of channels for this device. We support a possible
-+ // minimum device channel number > than the value requested by the user.
-+ stream_.nUserChannels[mode] = channels;
-+ unsigned int value;
-+ result = snd_pcm_hw_params_get_channels_max( hw_params, &value );
-+ unsigned int deviceChannels = value;
-+ if ( result < 0 || deviceChannels < channels + firstChannel ) {
-+ snd_pcm_close( phandle );
-+ errorStream_ << "RtApiAlsa::probeDeviceOpen: requested channel parameters not supported by device (" << name << "), " << snd_strerror( result ) << ".";
-+ errorText_ = errorStream_.str();
-+ return FAILURE;
-+ }
-+
-+ result = snd_pcm_hw_params_get_channels_min( hw_params, &value );
-+ if ( result < 0 ) {
-+ snd_pcm_close( phandle );
-+ errorStream_ << "RtApiAlsa::probeDeviceOpen: error getting minimum channels for device (" << name << "), " << snd_strerror( result ) << ".";
-+ errorText_ = errorStream_.str();
-+ return FAILURE;
-+ }
-+ deviceChannels = value;
-+ if ( deviceChannels < channels + firstChannel ) deviceChannels = channels + firstChannel;
-+ stream_.nDeviceChannels[mode] = deviceChannels;
-+
-+ // Set the device channels.
-+ result = snd_pcm_hw_params_set_channels( phandle, hw_params, deviceChannels );
-+ if ( result < 0 ) {
-+ snd_pcm_close( phandle );
-+ errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting channels for device (" << name << "), " << snd_strerror( result ) << ".";
-+ errorText_ = errorStream_.str();
-+ return FAILURE;
-+ }
-+
-+ // Set the buffer (or period) size.
-+ int dir = 0;
-+ snd_pcm_uframes_t periodSize = *bufferSize;
-+ result = snd_pcm_hw_params_set_period_size_near( phandle, hw_params, &periodSize, &dir );
-+ if ( result < 0 ) {
-+ snd_pcm_close( phandle );
-+ errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting period size for device (" << name << "), " << snd_strerror( result ) << ".";
-+ errorText_ = errorStream_.str();
-+ return FAILURE;
-+ }
-+ *bufferSize = periodSize;
-+
-+ // Set the buffer number, which in ALSA is referred to as the "period".
-+ unsigned int periods = 0;
-+ if ( options && options->flags & RTAUDIO_MINIMIZE_LATENCY ) periods = 2;
-+ if ( options && options->numberOfBuffers > 0 ) periods = options->numberOfBuffers;
-+ if ( periods < 2 ) periods = 4; // a fairly safe default value
-+ result = snd_pcm_hw_params_set_periods_near( phandle, hw_params, &periods, &dir );
-+ if ( result < 0 ) {
-+ snd_pcm_close( phandle );
-+ errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting periods for device (" << name << "), " << snd_strerror( result ) << ".";
-+ errorText_ = errorStream_.str();
-+ return FAILURE;
-+ }
-+
-+ // If attempting to setup a duplex stream, the bufferSize parameter
-+ // MUST be the same in both directions!
-+ if ( stream_.mode == OUTPUT && mode == INPUT && *bufferSize != stream_.bufferSize ) {
-+ snd_pcm_close( phandle );
-+ errorStream_ << "RtApiAlsa::probeDeviceOpen: system error setting buffer size for duplex stream on device (" << name << ").";
-+ errorText_ = errorStream_.str();
-+ return FAILURE;
-+ }
-+
-+ stream_.bufferSize = *bufferSize;
-+
-+ // Install the hardware configuration
-+ result = snd_pcm_hw_params( phandle, hw_params );
-+ if ( result < 0 ) {
-+ snd_pcm_close( phandle );
-+ errorStream_ << "RtApiAlsa::probeDeviceOpen: error installing hardware configuration on device (" << name << "), " << snd_strerror( result ) << ".";
-+ errorText_ = errorStream_.str();
-+ return FAILURE;
-+ }
-+
-+#if defined(__RTAUDIO_DEBUG__)
-+ fprintf(stderr, "\nRtApiAlsa: dump hardware params after installation:\n\n");
-+ snd_pcm_hw_params_dump( hw_params, out );
-+#endif
-+
-+ // Set the software configuration to fill buffers with zeros and prevent device stopping on xruns.
-+ snd_pcm_sw_params_t *sw_params = NULL;
-+ snd_pcm_sw_params_alloca( &sw_params );
-+ snd_pcm_sw_params_current( phandle, sw_params );
-+ snd_pcm_sw_params_set_start_threshold( phandle, sw_params, *bufferSize );
-+ snd_pcm_sw_params_set_stop_threshold( phandle, sw_params, ULONG_MAX );
-+ snd_pcm_sw_params_set_silence_threshold( phandle, sw_params, 0 );
-+
-+ // The following two settings were suggested by Theo Veenker
-+ //snd_pcm_sw_params_set_avail_min( phandle, sw_params, *bufferSize );
-+ //snd_pcm_sw_params_set_xfer_align( phandle, sw_params, 1 );
-+
-+ // here are two options for a fix
-+ //snd_pcm_sw_params_set_silence_size( phandle, sw_params, ULONG_MAX );
-+ snd_pcm_uframes_t val;
-+ snd_pcm_sw_params_get_boundary( sw_params, &val );
-+ snd_pcm_sw_params_set_silence_size( phandle, sw_params, val );
-+
-+ result = snd_pcm_sw_params( phandle, sw_params );
-+ if ( result < 0 ) {
-+ snd_pcm_close( phandle );
-+ errorStream_ << "RtApiAlsa::probeDeviceOpen: error installing software configuration on device (" << name << "), " << snd_strerror( result ) << ".";
-+ errorText_ = errorStream_.str();
-+ return FAILURE;
-+ }
-+
-+#if defined(__RTAUDIO_DEBUG__)
-+ fprintf(stderr, "\nRtApiAlsa: dump software params after installation:\n\n");
-+ snd_pcm_sw_params_dump( sw_params, out );
-+#endif
-+
-+ // Set flags for buffer conversion
-+ stream_.doConvertBuffer[mode] = false;
-+ if ( stream_.userFormat != stream_.deviceFormat[mode] )
-+ stream_.doConvertBuffer[mode] = true;
-+ if ( stream_.nUserChannels[mode] < stream_.nDeviceChannels[mode] )
-+ stream_.doConvertBuffer[mode] = true;
-+ if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] &&
-+ stream_.nUserChannels[mode] > 1 )
-+ stream_.doConvertBuffer[mode] = true;
-+
-+ // Allocate the ApiHandle if necessary and then save.
-+ AlsaHandle *apiInfo = 0;
-+ if ( stream_.apiHandle == 0 ) {
-+ try {
-+ apiInfo = (AlsaHandle *) new AlsaHandle;
-+ }
-+ catch ( std::bad_alloc& ) {
-+ errorText_ = "RtApiAlsa::probeDeviceOpen: error allocating AlsaHandle memory.";
-+ goto error;
-+ }
-+
-+ if ( pthread_cond_init( &apiInfo->runnable_cv, NULL ) ) {
-+ errorText_ = "RtApiAlsa::probeDeviceOpen: error initializing pthread condition variable.";
-+ goto error;
-+ }
-+
-+ stream_.apiHandle = (void *) apiInfo;
-+ apiInfo->handles[0] = 0;
-+ apiInfo->handles[1] = 0;
-+ }
-+ else {
-+ apiInfo = (AlsaHandle *) stream_.apiHandle;
-+ }
-+ apiInfo->handles[mode] = phandle;
-+ phandle = 0;
-+
-+ // Allocate necessary internal buffers.
-+ unsigned long bufferBytes;
-+ bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat );
-+ stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 );
-+ if ( stream_.userBuffer[mode] == NULL ) {
-+ errorText_ = "RtApiAlsa::probeDeviceOpen: error allocating user buffer memory.";
-+ goto error;
-+ }
-+
-+ if ( stream_.doConvertBuffer[mode] ) {
-+
-+ bool makeBuffer = true;
-+ bufferBytes = stream_.nDeviceChannels[mode] * formatBytes( stream_.deviceFormat[mode] );
-+ if ( mode == INPUT ) {
-+ if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) {
-+ unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] );
-+ if ( bufferBytes <= bytesOut ) makeBuffer = false;
-+ }
-+ }
-+
-+ if ( makeBuffer ) {
-+ bufferBytes *= *bufferSize;
-+ if ( stream_.deviceBuffer ) free( stream_.deviceBuffer );
-+ stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 );
-+ if ( stream_.deviceBuffer == NULL ) {
-+ errorText_ = "RtApiAlsa::probeDeviceOpen: error allocating device buffer memory.";
-+ goto error;
-+ }
-+ }
-+ }
-+
-+ stream_.sampleRate = sampleRate;
-+ stream_.nBuffers = periods;
-+ stream_.device[mode] = device;
-+ stream_.state = STREAM_STOPPED;
-+
-+ // Setup the buffer conversion information structure.
-+ if ( stream_.doConvertBuffer[mode] ) setConvertInfo( mode, firstChannel );
-+
-+ // Setup thread if necessary.
-+ if ( stream_.mode == OUTPUT && mode == INPUT ) {
-+ // We had already set up an output stream.
-+ stream_.mode = DUPLEX;
-+ // Link the streams if possible.
-+ apiInfo->synchronized = false;
-+ if ( snd_pcm_link( apiInfo->handles[0], apiInfo->handles[1] ) == 0 )
-+ apiInfo->synchronized = true;
-+ else {
-+ errorText_ = "RtApiAlsa::probeDeviceOpen: unable to synchronize input and output devices.";
-+ error( RtAudioError::WARNING );
-+ }
-+ }
-+ else {
-+ stream_.mode = mode;
-+
-+ // Setup callback thread.
-+ stream_.callbackInfo.object = (void *) this;
-+
-+ // Set the thread attributes for joinable and realtime scheduling
-+ // priority (optional). The higher priority will only take affect
-+ // if the program is run as root or suid. Note, under Linux
-+ // processes with CAP_SYS_NICE privilege, a user can change
-+ // scheduling policy and priority (thus need not be root). See
-+ // POSIX "capabilities".
-+ pthread_attr_t attr;
-+ pthread_attr_init( &attr );
-+ pthread_attr_setdetachstate( &attr, PTHREAD_CREATE_JOINABLE );
-+
-+#ifdef SCHED_RR // Undefined with some OSes (eg: NetBSD 1.6.x with GNU Pthread)
-+ if ( options && options->flags & RTAUDIO_SCHEDULE_REALTIME ) {
-+ // We previously attempted to increase the audio callback priority
-+ // to SCHED_RR here via the attributes. However, while no errors
-+ // were reported in doing so, it did not work. So, now this is
-+ // done in the alsaCallbackHandler function.
-+ stream_.callbackInfo.doRealtime = true;
-+ int priority = options->priority;
-+ int min = sched_get_priority_min( SCHED_RR );
-+ int max = sched_get_priority_max( SCHED_RR );
-+ if ( priority < min ) priority = min;
-+ else if ( priority > max ) priority = max;
-+ stream_.callbackInfo.priority = priority;
-+ }
-+#endif
-+
-+ stream_.callbackInfo.isRunning = true;
-+ result = pthread_create( &stream_.callbackInfo.thread, &attr, alsaCallbackHandler, &stream_.callbackInfo );
-+ pthread_attr_destroy( &attr );
-+ if ( result ) {
-+ stream_.callbackInfo.isRunning = false;
-+ errorText_ = "RtApiAlsa::error creating callback thread!";
-+ goto error;
-+ }
-+ }
-+
-+ return SUCCESS;
-+
-+ error:
-+ if ( apiInfo ) {
-+ pthread_cond_destroy( &apiInfo->runnable_cv );
-+ if ( apiInfo->handles[0] ) snd_pcm_close( apiInfo->handles[0] );
-+ if ( apiInfo->handles[1] ) snd_pcm_close( apiInfo->handles[1] );
-+ delete apiInfo;
-+ stream_.apiHandle = 0;
-+ }
-+
-+ if ( phandle) snd_pcm_close( phandle );
-+
-+ for ( int i=0; i<2; i++ ) {
-+ if ( stream_.userBuffer[i] ) {
-+ free( stream_.userBuffer[i] );
-+ stream_.userBuffer[i] = 0;
-+ }
-+ }
-+
-+ if ( stream_.deviceBuffer ) {
-+ free( stream_.deviceBuffer );
-+ stream_.deviceBuffer = 0;
-+ }
-+
-+ stream_.state = STREAM_CLOSED;
-+ return FAILURE;
-+}
-+
-+void RtApiAlsa :: closeStream()
-+{
-+ if ( stream_.state == STREAM_CLOSED ) {
-+ errorText_ = "RtApiAlsa::closeStream(): no open stream to close!";
-+ error( RtAudioError::WARNING );
-+ return;
-+ }
-+
-+ AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle;
-+ stream_.callbackInfo.isRunning = false;
-+ MUTEX_LOCK( &stream_.mutex );
-+ if ( stream_.state == STREAM_STOPPED ) {
-+ apiInfo->runnable = true;
-+ pthread_cond_signal( &apiInfo->runnable_cv );
-+ }
-+ MUTEX_UNLOCK( &stream_.mutex );
-+ pthread_join( stream_.callbackInfo.thread, NULL );
-+
-+ if ( stream_.state == STREAM_RUNNING ) {
-+ stream_.state = STREAM_STOPPED;
-+ if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX )
-+ snd_pcm_drop( apiInfo->handles[0] );
-+ if ( stream_.mode == INPUT || stream_.mode == DUPLEX )
-+ snd_pcm_drop( apiInfo->handles[1] );
-+ }
-+
-+ if ( apiInfo ) {
-+ pthread_cond_destroy( &apiInfo->runnable_cv );
-+ if ( apiInfo->handles[0] ) snd_pcm_close( apiInfo->handles[0] );
-+ if ( apiInfo->handles[1] ) snd_pcm_close( apiInfo->handles[1] );
-+ delete apiInfo;
-+ stream_.apiHandle = 0;
-+ }
-+
-+ for ( int i=0; i<2; i++ ) {
-+ if ( stream_.userBuffer[i] ) {
-+ free( stream_.userBuffer[i] );
-+ stream_.userBuffer[i] = 0;
-+ }
-+ }
-+
-+ if ( stream_.deviceBuffer ) {
-+ free( stream_.deviceBuffer );
-+ stream_.deviceBuffer = 0;
-+ }
-+
-+ stream_.mode = UNINITIALIZED;
-+ stream_.state = STREAM_CLOSED;
-+}
-+
-+void RtApiAlsa :: startStream()
-+{
-+ // This method calls snd_pcm_prepare if the device isn't already in that state.
-+
-+ verifyStream();
-+ if ( stream_.state == STREAM_RUNNING ) {
-+ errorText_ = "RtApiAlsa::startStream(): the stream is already running!";
-+ error( RtAudioError::WARNING );
-+ return;
-+ }
-+
-+ MUTEX_LOCK( &stream_.mutex );
-+
-+ int result = 0;
-+ snd_pcm_state_t state;
-+ AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle;
-+ snd_pcm_t **handle = (snd_pcm_t **) apiInfo->handles;
-+ if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
-+ state = snd_pcm_state( handle[0] );
-+ if ( state != SND_PCM_STATE_PREPARED ) {
-+ result = snd_pcm_prepare( handle[0] );
-+ if ( result < 0 ) {
-+ errorStream_ << "RtApiAlsa::startStream: error preparing output pcm device, " << snd_strerror( result ) << ".";
-+ errorText_ = errorStream_.str();
-+ goto unlock;
-+ }
-+ }
-+ }
-+
-+ if ( ( stream_.mode == INPUT || stream_.mode == DUPLEX ) && !apiInfo->synchronized ) {
-+ result = snd_pcm_drop(handle[1]); // fix to remove stale data received since device has been open
-+ state = snd_pcm_state( handle[1] );
-+ if ( state != SND_PCM_STATE_PREPARED ) {
-+ result = snd_pcm_prepare( handle[1] );
-+ if ( result < 0 ) {
-+ errorStream_ << "RtApiAlsa::startStream: error preparing input pcm device, " << snd_strerror( result ) << ".";
-+ errorText_ = errorStream_.str();
-+ goto unlock;
-+ }
-+ }
-+ }
-+
-+ stream_.state = STREAM_RUNNING;
-+
-+ unlock:
-+ apiInfo->runnable = true;
-+ pthread_cond_signal( &apiInfo->runnable_cv );
-+ MUTEX_UNLOCK( &stream_.mutex );
-+
-+ if ( result >= 0 ) return;
-+ error( RtAudioError::SYSTEM_ERROR );
-+}
-+
-+void RtApiAlsa :: stopStream()
-+{
-+ verifyStream();
-+ if ( stream_.state == STREAM_STOPPED ) {
-+ errorText_ = "RtApiAlsa::stopStream(): the stream is already stopped!";
-+ error( RtAudioError::WARNING );
-+ return;
-+ }
-+
-+ stream_.state = STREAM_STOPPED;
-+ MUTEX_LOCK( &stream_.mutex );
-+
-+ int result = 0;
-+ AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle;
-+ snd_pcm_t **handle = (snd_pcm_t **) apiInfo->handles;
-+ if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
-+ if ( apiInfo->synchronized )
-+ result = snd_pcm_drop( handle[0] );
-+ else
-+ result = snd_pcm_drain( handle[0] );
-+ if ( result < 0 ) {
-+ errorStream_ << "RtApiAlsa::stopStream: error draining output pcm device, " << snd_strerror( result ) << ".";
-+ errorText_ = errorStream_.str();
-+ goto unlock;
-+ }
-+ }
-+
-+ if ( ( stream_.mode == INPUT || stream_.mode == DUPLEX ) && !apiInfo->synchronized ) {
-+ result = snd_pcm_drop( handle[1] );
-+ if ( result < 0 ) {
-+ errorStream_ << "RtApiAlsa::stopStream: error stopping input pcm device, " << snd_strerror( result ) << ".";
-+ errorText_ = errorStream_.str();
-+ goto unlock;
-+ }
-+ }
-+
-+ unlock:
-+ apiInfo->runnable = false; // fixes high CPU usage when stopped
-+ MUTEX_UNLOCK( &stream_.mutex );
-+
-+ if ( result >= 0 ) return;
-+ error( RtAudioError::SYSTEM_ERROR );
-+}
-+
-+void RtApiAlsa :: abortStream()
-+{
-+ verifyStream();
-+ if ( stream_.state == STREAM_STOPPED ) {
-+ errorText_ = "RtApiAlsa::abortStream(): the stream is already stopped!";
-+ error( RtAudioError::WARNING );
-+ return;
-+ }
-+
-+ stream_.state = STREAM_STOPPED;
-+ MUTEX_LOCK( &stream_.mutex );
-+
-+ int result = 0;
-+ AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle;
-+ snd_pcm_t **handle = (snd_pcm_t **) apiInfo->handles;
-+ if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
-+ result = snd_pcm_drop( handle[0] );
-+ if ( result < 0 ) {
-+ errorStream_ << "RtApiAlsa::abortStream: error aborting output pcm device, " << snd_strerror( result ) << ".";
-+ errorText_ = errorStream_.str();
-+ goto unlock;
-+ }
-+ }
-+
-+ if ( ( stream_.mode == INPUT || stream_.mode == DUPLEX ) && !apiInfo->synchronized ) {
-+ result = snd_pcm_drop( handle[1] );
-+ if ( result < 0 ) {
-+ errorStream_ << "RtApiAlsa::abortStream: error aborting input pcm device, " << snd_strerror( result ) << ".";
-+ errorText_ = errorStream_.str();
-+ goto unlock;
-+ }
-+ }
-+
-+ unlock:
-+ apiInfo->runnable = false; // fixes high CPU usage when stopped
-+ MUTEX_UNLOCK( &stream_.mutex );
-+
-+ if ( result >= 0 ) return;
-+ error( RtAudioError::SYSTEM_ERROR );
-+}
-+
-+void RtApiAlsa :: callbackEvent()
-+{
-+ AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle;
-+ if ( stream_.state == STREAM_STOPPED ) {
-+ MUTEX_LOCK( &stream_.mutex );
-+ while ( !apiInfo->runnable )
-+ pthread_cond_wait( &apiInfo->runnable_cv, &stream_.mutex );
-+
-+ if ( stream_.state != STREAM_RUNNING ) {
-+ MUTEX_UNLOCK( &stream_.mutex );
-+ return;
-+ }
-+ MUTEX_UNLOCK( &stream_.mutex );
-+ }
-+
-+ if ( stream_.state == STREAM_CLOSED ) {
-+ errorText_ = "RtApiAlsa::callbackEvent(): the stream is closed ... this shouldn't happen!";
-+ error( RtAudioError::WARNING );
-+ return;
-+ }
-+
-+ int doStopStream = 0;
-+ RtAudioCallback callback = (RtAudioCallback) stream_.callbackInfo.callback;
-+ double streamTime = getStreamTime();
-+ RtAudioStreamStatus status = 0;
-+ if ( stream_.mode != INPUT && apiInfo->xrun[0] == true ) {
-+ status |= RTAUDIO_OUTPUT_UNDERFLOW;
-+ apiInfo->xrun[0] = false;
-+ }
-+ if ( stream_.mode != OUTPUT && apiInfo->xrun[1] == true ) {
-+ status |= RTAUDIO_INPUT_OVERFLOW;
-+ apiInfo->xrun[1] = false;
-+ }
-+ doStopStream = callback( stream_.userBuffer[0], stream_.userBuffer[1],
-+ stream_.bufferSize, streamTime, status, stream_.callbackInfo.userData );
-+
-+ if ( doStopStream == 2 ) {
-+ abortStream();
-+ return;
-+ }
-+
-+ MUTEX_LOCK( &stream_.mutex );
-+
-+ // The state might change while waiting on a mutex.
-+ if ( stream_.state == STREAM_STOPPED ) goto unlock;
-+
-+ int result;
-+ char *buffer;
-+ int channels;
-+ snd_pcm_t **handle;
-+ snd_pcm_sframes_t frames;
-+ RtAudioFormat format;
-+ handle = (snd_pcm_t **) apiInfo->handles;
-+
-+ if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {
-+
-+ // Setup parameters.
-+ if ( stream_.doConvertBuffer[1] ) {
-+ buffer = stream_.deviceBuffer;
-+ channels = stream_.nDeviceChannels[1];
-+ format = stream_.deviceFormat[1];
-+ }
-+ else {
-+ buffer = stream_.userBuffer[1];
-+ channels = stream_.nUserChannels[1];
-+ format = stream_.userFormat;
-+ }
-+
-+ // Read samples from device in interleaved/non-interleaved format.
-+ if ( stream_.deviceInterleaved[1] )
-+ result = snd_pcm_readi( handle[1], buffer, stream_.bufferSize );
-+ else {
-+ void *bufs[channels];
-+ size_t offset = stream_.bufferSize * formatBytes( format );
-+ for ( int i=0; i<channels; i++ )
-+ bufs[i] = (void *) (buffer + (i * offset));
-+ result = snd_pcm_readn( handle[1], bufs, stream_.bufferSize );
-+ }
-+
-+ if ( result < (int) stream_.bufferSize ) {
-+ // Either an error or overrun occured.
-+ if ( result == -EPIPE ) {
-+ snd_pcm_state_t state = snd_pcm_state( handle[1] );
-+ if ( state == SND_PCM_STATE_XRUN ) {
-+ apiInfo->xrun[1] = true;
-+ result = snd_pcm_prepare( handle[1] );
-+ if ( result < 0 ) {
-+ errorStream_ << "RtApiAlsa::callbackEvent: error preparing device after overrun, " << snd_strerror( result ) << ".";
-+ errorText_ = errorStream_.str();
-+ }
-+ }
-+ else {
-+ errorStream_ << "RtApiAlsa::callbackEvent: error, current state is " << snd_pcm_state_name( state ) << ", " << snd_strerror( result ) << ".";
-+ errorText_ = errorStream_.str();
-+ }
-+ }
-+ else {
-+ errorStream_ << "RtApiAlsa::callbackEvent: audio read error, " << snd_strerror( result ) << ".";
-+ errorText_ = errorStream_.str();
-+ }
-+ error( RtAudioError::WARNING );
-+ goto tryOutput;
-+ }
-+
-+ // Do byte swapping if necessary.
-+ if ( stream_.doByteSwap[1] )
-+ byteSwapBuffer( buffer, stream_.bufferSize * channels, format );
-+
-+ // Do buffer conversion if necessary.
-+ if ( stream_.doConvertBuffer[1] )
-+ convertBuffer( stream_.userBuffer[1], stream_.deviceBuffer, stream_.convertInfo[1] );
-+
-+ // Check stream latency
-+ result = snd_pcm_delay( handle[1], &frames );
-+ if ( result == 0 && frames > 0 ) stream_.latency[1] = frames;
-+ }
-+
-+ tryOutput:
-+
-+ if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
-+
-+ // Setup parameters and do buffer conversion if necessary.
-+ if ( stream_.doConvertBuffer[0] ) {
-+ buffer = stream_.deviceBuffer;
-+ convertBuffer( buffer, stream_.userBuffer[0], stream_.convertInfo[0] );
-+ channels = stream_.nDeviceChannels[0];
-+ format = stream_.deviceFormat[0];
-+ }
-+ else {
-+ buffer = stream_.userBuffer[0];
-+ channels = stream_.nUserChannels[0];
-+ format = stream_.userFormat;
-+ }
-+
-+ // Do byte swapping if necessary.
-+ if ( stream_.doByteSwap[0] )
-+ byteSwapBuffer(buffer, stream_.bufferSize * channels, format);
-+
-+ // Write samples to device in interleaved/non-interleaved format.
-+ if ( stream_.deviceInterleaved[0] )
-+ result = snd_pcm_writei( handle[0], buffer, stream_.bufferSize );
-+ else {
-+ void *bufs[channels];
-+ size_t offset = stream_.bufferSize * formatBytes( format );
-+ for ( int i=0; i<channels; i++ )
-+ bufs[i] = (void *) (buffer + (i * offset));
-+ result = snd_pcm_writen( handle[0], bufs, stream_.bufferSize );
-+ }
-+
-+ if ( result < (int) stream_.bufferSize ) {
-+ // Either an error or underrun occured.
-+ if ( result == -EPIPE ) {
-+ snd_pcm_state_t state = snd_pcm_state( handle[0] );
-+ if ( state == SND_PCM_STATE_XRUN ) {
-+ apiInfo->xrun[0] = true;
-+ result = snd_pcm_prepare( handle[0] );
-+ if ( result < 0 ) {
-+ errorStream_ << "RtApiAlsa::callbackEvent: error preparing device after underrun, " << snd_strerror( result ) << ".";
-+ errorText_ = errorStream_.str();
-+ }
-+ else
-+ errorText_ = "RtApiAlsa::callbackEvent: audio write error, underrun.";
-+ }
-+ else {
-+ errorStream_ << "RtApiAlsa::callbackEvent: error, current state is " << snd_pcm_state_name( state ) << ", " << snd_strerror( result ) << ".";
-+ errorText_ = errorStream_.str();
-+ }
-+ }
-+ else {
-+ errorStream_ << "RtApiAlsa::callbackEvent: audio write error, " << snd_strerror( result ) << ".";
-+ errorText_ = errorStream_.str();
-+ }
-+ error( RtAudioError::WARNING );
-+ goto unlock;
-+ }
-+
-+ // Check stream latency
-+ result = snd_pcm_delay( handle[0], &frames );
-+ if ( result == 0 && frames > 0 ) stream_.latency[0] = frames;
-+ }
-+
-+ unlock:
-+ MUTEX_UNLOCK( &stream_.mutex );
-+
-+ RtApi::tickStreamTime();
-+ if ( doStopStream == 1 ) this->stopStream();
-+}
-+
-+static void *alsaCallbackHandler( void *ptr )
-+{
-+ CallbackInfo *info = (CallbackInfo *) ptr;
-+ RtApiAlsa *object = (RtApiAlsa *) info->object;
-+ bool *isRunning = &info->isRunning;
-+
-+#ifdef SCHED_RR // Undefined with some OSes (eg: NetBSD 1.6.x with GNU Pthread)
-+ if ( info->doRealtime ) {
-+ pthread_t tID = pthread_self(); // ID of this thread
-+ sched_param prio = { info->priority }; // scheduling priority of thread
-+ pthread_setschedparam( tID, SCHED_RR, &prio );
-+ }
-+#endif
-+
-+ while ( *isRunning == true ) {
-+ pthread_testcancel();
-+ object->callbackEvent();
-+ }
-+
-+ pthread_exit( NULL );
-+}
-+
-+//******************** End of __LINUX_ALSA__ *********************//
-+#endif
-+
-+#if defined(__LINUX_PULSE__)
-+
-+// Code written by Peter Meerwald, pmeerw@pmeerw.net
-+// and Tristan Matthews.
-+
-+#include <pulse/error.h>
-+#include <pulse/simple.h>
-+#include <cstdio>
-+
-+static const unsigned int SUPPORTED_SAMPLERATES[] = { 8000, 16000, 22050, 32000,
-+ 44100, 48000, 96000, 0};
-+
-+struct rtaudio_pa_format_mapping_t {
-+ RtAudioFormat rtaudio_format;
-+ pa_sample_format_t pa_format;
-+};
-+
-+static const rtaudio_pa_format_mapping_t supported_sampleformats[] = {
-+ {RTAUDIO_SINT16, PA_SAMPLE_S16LE},
-+ {RTAUDIO_SINT32, PA_SAMPLE_S32LE},
-+ {RTAUDIO_FLOAT32, PA_SAMPLE_FLOAT32LE},
-+ {0, PA_SAMPLE_INVALID}};
-+
-+struct PulseAudioHandle {
-+ pa_simple *s_play;
-+ pa_simple *s_rec;
-+ pthread_t thread;
-+ pthread_cond_t runnable_cv;
-+ bool runnable;
-+ PulseAudioHandle() : s_play(0), s_rec(0), runnable(false) { }
-+};
-+
-+RtApiPulse::~RtApiPulse()
-+{
-+ if ( stream_.state != STREAM_CLOSED )
-+ closeStream();
-+}
-+
-+unsigned int RtApiPulse::getDeviceCount( void )
-+{
-+ return 1;
-+}
-+
-+RtAudio::DeviceInfo RtApiPulse::getDeviceInfo( unsigned int /*device*/ )
-+{
-+ RtAudio::DeviceInfo info;
-+ info.probed = true;
-+ info.name = "PulseAudio";
-+ info.outputChannels = 2;
-+ info.inputChannels = 2;
-+ info.duplexChannels = 2;
-+ info.isDefaultOutput = true;
-+ info.isDefaultInput = true;
-+
-+ for ( const unsigned int *sr = SUPPORTED_SAMPLERATES; *sr; ++sr )
-+ info.sampleRates.push_back( *sr );
-+
-+ info.preferredSampleRate = 48000;
-+ info.nativeFormats = RTAUDIO_SINT16 | RTAUDIO_SINT32 | RTAUDIO_FLOAT32;
-+
-+ return info;
-+}
-+
-+static void *pulseaudio_callback( void * user )
-+{
-+ CallbackInfo *cbi = static_cast<CallbackInfo *>( user );
-+ RtApiPulse *context = static_cast<RtApiPulse *>( cbi->object );
-+ volatile bool *isRunning = &cbi->isRunning;
-+
-+ while ( *isRunning ) {
-+ pthread_testcancel();
-+ context->callbackEvent();
-+ }
-+
-+ pthread_exit( NULL );
-+}
-+
-+void RtApiPulse::closeStream( void )
-+{
-+ PulseAudioHandle *pah = static_cast<PulseAudioHandle *>( stream_.apiHandle );
-+
-+ stream_.callbackInfo.isRunning = false;
-+ if ( pah ) {
-+ MUTEX_LOCK( &stream_.mutex );
-+ if ( stream_.state == STREAM_STOPPED ) {
-+ pah->runnable = true;
-+ pthread_cond_signal( &pah->runnable_cv );
-+ }
-+ MUTEX_UNLOCK( &stream_.mutex );
-+
-+ pthread_join( pah->thread, 0 );
-+ if ( pah->s_play ) {
-+ pa_simple_flush( pah->s_play, NULL );
-+ pa_simple_free( pah->s_play );
-+ }
-+ if ( pah->s_rec )
-+ pa_simple_free( pah->s_rec );
-+
-+ pthread_cond_destroy( &pah->runnable_cv );
-+ delete pah;
-+ stream_.apiHandle = 0;
-+ }
-+
-+ if ( stream_.userBuffer[0] ) {
-+ free( stream_.userBuffer[0] );
-+ stream_.userBuffer[0] = 0;
-+ }
-+ if ( stream_.userBuffer[1] ) {
-+ free( stream_.userBuffer[1] );
-+ stream_.userBuffer[1] = 0;
-+ }
-+
-+ stream_.state = STREAM_CLOSED;
-+ stream_.mode = UNINITIALIZED;
-+}
-+
-+void RtApiPulse::callbackEvent( void )
-+{
-+ PulseAudioHandle *pah = static_cast<PulseAudioHandle *>( stream_.apiHandle );
-+
-+ if ( stream_.state == STREAM_STOPPED ) {
-+ MUTEX_LOCK( &stream_.mutex );
-+ while ( !pah->runnable )
-+ pthread_cond_wait( &pah->runnable_cv, &stream_.mutex );
-+
-+ if ( stream_.state != STREAM_RUNNING ) {
-+ MUTEX_UNLOCK( &stream_.mutex );
-+ return;
-+ }
-+ MUTEX_UNLOCK( &stream_.mutex );
-+ }
-+
-+ if ( stream_.state == STREAM_CLOSED ) {
-+ errorText_ = "RtApiPulse::callbackEvent(): the stream is closed ... "
-+ "this shouldn't happen!";
-+ error( RtAudioError::WARNING );
-+ return;
-+ }
-+
-+ RtAudioCallback callback = (RtAudioCallback) stream_.callbackInfo.callback;
-+ double streamTime = getStreamTime();
-+ RtAudioStreamStatus status = 0;
-+ int doStopStream = callback( stream_.userBuffer[OUTPUT], stream_.userBuffer[INPUT],
-+ stream_.bufferSize, streamTime, status,
-+ stream_.callbackInfo.userData );
-+
-+ if ( doStopStream == 2 ) {
-+ abortStream();
-+ return;
-+ }
-+
-+ MUTEX_LOCK( &stream_.mutex );
-+ void *pulse_in = stream_.doConvertBuffer[INPUT] ? stream_.deviceBuffer : stream_.userBuffer[INPUT];
-+ void *pulse_out = stream_.doConvertBuffer[OUTPUT] ? stream_.deviceBuffer : stream_.userBuffer[OUTPUT];
-+
-+ if ( stream_.state != STREAM_RUNNING )
-+ goto unlock;
-+
-+ int pa_error;
-+ size_t bytes;
-+ if (stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
-+ if ( stream_.doConvertBuffer[OUTPUT] ) {
-+ convertBuffer( stream_.deviceBuffer,
-+ stream_.userBuffer[OUTPUT],
-+ stream_.convertInfo[OUTPUT] );
-+ bytes = stream_.nDeviceChannels[OUTPUT] * stream_.bufferSize *
-+ formatBytes( stream_.deviceFormat[OUTPUT] );
-+ } else
-+ bytes = stream_.nUserChannels[OUTPUT] * stream_.bufferSize *
-+ formatBytes( stream_.userFormat );
-+
-+ if ( pa_simple_write( pah->s_play, pulse_out, bytes, &pa_error ) < 0 ) {
-+ errorStream_ << "RtApiPulse::callbackEvent: audio write error, " <<
-+ pa_strerror( pa_error ) << ".";
-+ errorText_ = errorStream_.str();
-+ error( RtAudioError::WARNING );
-+ }
-+ }
-+
-+ if ( stream_.mode == INPUT || stream_.mode == DUPLEX) {
-+ if ( stream_.doConvertBuffer[INPUT] )
-+ bytes = stream_.nDeviceChannels[INPUT] * stream_.bufferSize *
-+ formatBytes( stream_.deviceFormat[INPUT] );
-+ else
-+ bytes = stream_.nUserChannels[INPUT] * stream_.bufferSize *
-+ formatBytes( stream_.userFormat );
-+
-+ if ( pa_simple_read( pah->s_rec, pulse_in, bytes, &pa_error ) < 0 ) {
-+ errorStream_ << "RtApiPulse::callbackEvent: audio read error, " <<
-+ pa_strerror( pa_error ) << ".";
-+ errorText_ = errorStream_.str();
-+ error( RtAudioError::WARNING );
-+ }
-+ if ( stream_.doConvertBuffer[INPUT] ) {
-+ convertBuffer( stream_.userBuffer[INPUT],
-+ stream_.deviceBuffer,
-+ stream_.convertInfo[INPUT] );
-+ }
-+ }
-+
-+ unlock:
-+ MUTEX_UNLOCK( &stream_.mutex );
-+ RtApi::tickStreamTime();
-+
-+ if ( doStopStream == 1 )
-+ stopStream();
-+}
-+
-+void RtApiPulse::startStream( void )
-+{
-+ PulseAudioHandle *pah = static_cast<PulseAudioHandle *>( stream_.apiHandle );
-+
-+ if ( stream_.state == STREAM_CLOSED ) {
-+ errorText_ = "RtApiPulse::startStream(): the stream is not open!";
-+ error( RtAudioError::INVALID_USE );
-+ return;
-+ }
-+ if ( stream_.state == STREAM_RUNNING ) {
-+ errorText_ = "RtApiPulse::startStream(): the stream is already running!";
-+ error( RtAudioError::WARNING );
-+ return;
-+ }
-+
-+ MUTEX_LOCK( &stream_.mutex );
-+
-+ stream_.state = STREAM_RUNNING;
-+
-+ pah->runnable = true;
-+ pthread_cond_signal( &pah->runnable_cv );
-+ MUTEX_UNLOCK( &stream_.mutex );
-+}
-+
-+void RtApiPulse::stopStream( void )
-+{
-+ PulseAudioHandle *pah = static_cast<PulseAudioHandle *>( stream_.apiHandle );
-+
-+ if ( stream_.state == STREAM_CLOSED ) {
-+ errorText_ = "RtApiPulse::stopStream(): the stream is not open!";
-+ error( RtAudioError::INVALID_USE );
-+ return;
-+ }
-+ if ( stream_.state == STREAM_STOPPED ) {
-+ errorText_ = "RtApiPulse::stopStream(): the stream is already stopped!";
-+ error( RtAudioError::WARNING );
-+ return;
-+ }
-+
-+ stream_.state = STREAM_STOPPED;
-+ MUTEX_LOCK( &stream_.mutex );
-+
-+ if ( pah && pah->s_play ) {
-+ int pa_error;
-+ if ( pa_simple_drain( pah->s_play, &pa_error ) < 0 ) {
-+ errorStream_ << "RtApiPulse::stopStream: error draining output device, " <<
-+ pa_strerror( pa_error ) << ".";
-+ errorText_ = errorStream_.str();
-+ MUTEX_UNLOCK( &stream_.mutex );
-+ error( RtAudioError::SYSTEM_ERROR );
-+ return;
-+ }
-+ }
-+
-+ stream_.state = STREAM_STOPPED;
-+ MUTEX_UNLOCK( &stream_.mutex );
-+}
-+
-+void RtApiPulse::abortStream( void )
-+{
-+ PulseAudioHandle *pah = static_cast<PulseAudioHandle*>( stream_.apiHandle );
-+
-+ if ( stream_.state == STREAM_CLOSED ) {
-+ errorText_ = "RtApiPulse::abortStream(): the stream is not open!";
-+ error( RtAudioError::INVALID_USE );
-+ return;
-+ }
-+ if ( stream_.state == STREAM_STOPPED ) {
-+ errorText_ = "RtApiPulse::abortStream(): the stream is already stopped!";
-+ error( RtAudioError::WARNING );
-+ return;
-+ }
-+
-+ stream_.state = STREAM_STOPPED;
-+ MUTEX_LOCK( &stream_.mutex );
-+
-+ if ( pah && pah->s_play ) {
-+ int pa_error;
-+ if ( pa_simple_flush( pah->s_play, &pa_error ) < 0 ) {
-+ errorStream_ << "RtApiPulse::abortStream: error flushing output device, " <<
-+ pa_strerror( pa_error ) << ".";
-+ errorText_ = errorStream_.str();
-+ MUTEX_UNLOCK( &stream_.mutex );
-+ error( RtAudioError::SYSTEM_ERROR );
-+ return;
-+ }
-+ }
-+
-+ stream_.state = STREAM_STOPPED;
-+ MUTEX_UNLOCK( &stream_.mutex );
-+}
-+
-+bool RtApiPulse::probeDeviceOpen( unsigned int device, StreamMode mode,
-+ unsigned int channels, unsigned int firstChannel,
-+ unsigned int sampleRate, RtAudioFormat format,
-+ unsigned int *bufferSize, RtAudio::StreamOptions *options )
-+{
-+ PulseAudioHandle *pah = 0;
-+ unsigned long bufferBytes = 0;
-+ pa_sample_spec ss;
-+
-+ if ( device != 0 ) return false;
-+ if ( mode != INPUT && mode != OUTPUT ) return false;
-+ if ( channels != 1 && channels != 2 ) {
-+ errorText_ = "RtApiPulse::probeDeviceOpen: unsupported number of channels.";
-+ return false;
-+ }
-+ ss.channels = channels;
-+
-+ if ( firstChannel != 0 ) return false;
-+
-+ bool sr_found = false;
-+ for ( const unsigned int *sr = SUPPORTED_SAMPLERATES; *sr; ++sr ) {
-+ if ( sampleRate == *sr ) {
-+ sr_found = true;
-+ stream_.sampleRate = sampleRate;
-+ ss.rate = sampleRate;
-+ break;
-+ }
-+ }
-+ if ( !sr_found ) {
-+ errorText_ = "RtApiPulse::probeDeviceOpen: unsupported sample rate.";
-+ return false;
-+ }
-+
-+ bool sf_found = 0;
-+ for ( const rtaudio_pa_format_mapping_t *sf = supported_sampleformats;
-+ sf->rtaudio_format && sf->pa_format != PA_SAMPLE_INVALID; ++sf ) {
-+ if ( format == sf->rtaudio_format ) {
-+ sf_found = true;
-+ stream_.userFormat = sf->rtaudio_format;
-+ stream_.deviceFormat[mode] = stream_.userFormat;
-+ ss.format = sf->pa_format;
-+ break;
-+ }
-+ }
-+ if ( !sf_found ) { // Use internal data format conversion.
-+ stream_.userFormat = format;
-+ stream_.deviceFormat[mode] = RTAUDIO_FLOAT32;
-+ ss.format = PA_SAMPLE_FLOAT32LE;
-+ }
-+
-+ // Set other stream parameters.
-+ if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) stream_.userInterleaved = false;
-+ else stream_.userInterleaved = true;
-+ stream_.deviceInterleaved[mode] = true;
-+ stream_.nBuffers = 1;
-+ stream_.doByteSwap[mode] = false;
-+ stream_.nUserChannels[mode] = channels;
-+ stream_.nDeviceChannels[mode] = channels + firstChannel;
-+ stream_.channelOffset[mode] = 0;
-+ std::string streamName = "RtAudio";
-+
-+ // Set flags for buffer conversion.
-+ stream_.doConvertBuffer[mode] = false;
-+ if ( stream_.userFormat != stream_.deviceFormat[mode] )
-+ stream_.doConvertBuffer[mode] = true;
-+ if ( stream_.nUserChannels[mode] < stream_.nDeviceChannels[mode] )
-+ stream_.doConvertBuffer[mode] = true;
-+
-+ // Allocate necessary internal buffers.
-+ bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat );
-+ stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 );
-+ if ( stream_.userBuffer[mode] == NULL ) {
-+ errorText_ = "RtApiPulse::probeDeviceOpen: error allocating user buffer memory.";
-+ goto error;
-+ }
-+ stream_.bufferSize = *bufferSize;
-+
-+ if ( stream_.doConvertBuffer[mode] ) {
-+
-+ bool makeBuffer = true;
-+ bufferBytes = stream_.nDeviceChannels[mode] * formatBytes( stream_.deviceFormat[mode] );
-+ if ( mode == INPUT ) {
-+ if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) {
-+ unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] );
-+ if ( bufferBytes <= bytesOut ) makeBuffer = false;
-+ }
-+ }
-+
-+ if ( makeBuffer ) {
-+ bufferBytes *= *bufferSize;
-+ if ( stream_.deviceBuffer ) free( stream_.deviceBuffer );
-+ stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 );
-+ if ( stream_.deviceBuffer == NULL ) {
-+ errorText_ = "RtApiPulse::probeDeviceOpen: error allocating device buffer memory.";
-+ goto error;
-+ }
-+ }
-+ }
-+
-+ stream_.device[mode] = device;
-+
-+ // Setup the buffer conversion information structure.
-+ if ( stream_.doConvertBuffer[mode] ) setConvertInfo( mode, firstChannel );
-+
-+ if ( !stream_.apiHandle ) {
-+ PulseAudioHandle *pah = new PulseAudioHandle;
-+ if ( !pah ) {
-+ errorText_ = "RtApiPulse::probeDeviceOpen: error allocating memory for handle.";
-+ goto error;
-+ }
-+
-+ stream_.apiHandle = pah;
-+ if ( pthread_cond_init( &pah->runnable_cv, NULL ) != 0 ) {
-+ errorText_ = "RtApiPulse::probeDeviceOpen: error creating condition variable.";
-+ goto error;
-+ }
-+ }
-+ pah = static_cast<PulseAudioHandle *>( stream_.apiHandle );
-+
-+ int error;
-+ if ( options && !options->streamName.empty() ) streamName = options->streamName;
-+ switch ( mode ) {
-+ case INPUT:
-+ pa_buffer_attr buffer_attr;
-+ buffer_attr.fragsize = bufferBytes;
-+ buffer_attr.maxlength = -1;
-+
-+ pah->s_rec = pa_simple_new( NULL, streamName.c_str(), PA_STREAM_RECORD, NULL, "Record", &ss, NULL, &buffer_attr, &error );
-+ if ( !pah->s_rec ) {
-+ errorText_ = "RtApiPulse::probeDeviceOpen: error connecting input to PulseAudio server.";
-+ goto error;
-+ }
-+ break;
-+ case OUTPUT:
-+ pah->s_play = pa_simple_new( NULL, streamName.c_str(), PA_STREAM_PLAYBACK, NULL, "Playback", &ss, NULL, NULL, &error );
-+ if ( !pah->s_play ) {
-+ errorText_ = "RtApiPulse::probeDeviceOpen: error connecting output to PulseAudio server.";
-+ goto error;
-+ }
-+ break;
-+ default:
-+ goto error;
-+ }
-+
-+ if ( stream_.mode == UNINITIALIZED )
-+ stream_.mode = mode;
-+ else if ( stream_.mode == mode )
-+ goto error;
-+ else
-+ stream_.mode = DUPLEX;
-+
-+ if ( !stream_.callbackInfo.isRunning ) {
-+ stream_.callbackInfo.object = this;
-+ stream_.callbackInfo.isRunning = true;
-+ if ( pthread_create( &pah->thread, NULL, pulseaudio_callback, (void *)&stream_.callbackInfo) != 0 ) {
-+ errorText_ = "RtApiPulse::probeDeviceOpen: error creating thread.";
-+ goto error;
-+ }
-+ }
-+
-+ stream_.state = STREAM_STOPPED;
-+ return true;
-+
-+ error:
-+ if ( pah && stream_.callbackInfo.isRunning ) {
-+ pthread_cond_destroy( &pah->runnable_cv );
-+ delete pah;
-+ stream_.apiHandle = 0;
-+ }
-+
-+ for ( int i=0; i<2; i++ ) {
-+ if ( stream_.userBuffer[i] ) {
-+ free( stream_.userBuffer[i] );
-+ stream_.userBuffer[i] = 0;
-+ }
-+ }
-+
-+ if ( stream_.deviceBuffer ) {
-+ free( stream_.deviceBuffer );
-+ stream_.deviceBuffer = 0;
-+ }
-+
-+ return FAILURE;
-+}
-+
-+//******************** End of __LINUX_PULSE__ *********************//
-+#endif
-+
-+#if defined(__LINUX_OSS__)
-+
-+#include <unistd.h>
-+#include <sys/ioctl.h>
-+#include <unistd.h>
-+#include <fcntl.h>
-+#include <sys/soundcard.h>
-+#include <errno.h>
-+#include <math.h>
-+
-+static void *ossCallbackHandler(void * ptr);
-+
-+// A structure to hold various information related to the OSS API
-+// implementation.
-+struct OssHandle {
-+ int id[2]; // device ids
-+ bool xrun[2];
-+ bool triggered;
-+ pthread_cond_t runnable;
-+
-+ OssHandle()
-+ :triggered(false) { id[0] = 0; id[1] = 0; xrun[0] = false; xrun[1] = false; }
-+};
-+
-+RtApiOss :: RtApiOss()
-+{
-+ // Nothing to do here.
-+}
-+
-+RtApiOss :: ~RtApiOss()
-+{
-+ if ( stream_.state != STREAM_CLOSED ) closeStream();
-+}
-+
-+unsigned int RtApiOss :: getDeviceCount( void )
-+{
-+ int mixerfd = open( "/dev/mixer", O_RDWR, 0 );
-+ if ( mixerfd == -1 ) {
-+ errorText_ = "RtApiOss::getDeviceCount: error opening '/dev/mixer'.";
-+ error( RtAudioError::WARNING );
-+ return 0;
-+ }
-+
-+ oss_sysinfo sysinfo;
-+ if ( ioctl( mixerfd, SNDCTL_SYSINFO, &sysinfo ) == -1 ) {
-+ close( mixerfd );
-+ errorText_ = "RtApiOss::getDeviceCount: error getting sysinfo, OSS version >= 4.0 is required.";
-+ error( RtAudioError::WARNING );
-+ return 0;
-+ }
-+
-+ close( mixerfd );
-+ return sysinfo.numaudios;
-+}
-+
-+RtAudio::DeviceInfo RtApiOss :: getDeviceInfo( unsigned int device )
-+{
-+ RtAudio::DeviceInfo info;
-+ info.probed = false;
-+
-+ int mixerfd = open( "/dev/mixer", O_RDWR, 0 );
-+ if ( mixerfd == -1 ) {
-+ errorText_ = "RtApiOss::getDeviceInfo: error opening '/dev/mixer'.";
-+ error( RtAudioError::WARNING );
-+ return info;
-+ }
-+
-+ oss_sysinfo sysinfo;
-+ int result = ioctl( mixerfd, SNDCTL_SYSINFO, &sysinfo );
-+ if ( result == -1 ) {
-+ close( mixerfd );
-+ errorText_ = "RtApiOss::getDeviceInfo: error getting sysinfo, OSS version >= 4.0 is required.";
-+ error( RtAudioError::WARNING );
-+ return info;
-+ }
-+
-+ unsigned nDevices = sysinfo.numaudios;
-+ if ( nDevices == 0 ) {
-+ close( mixerfd );
-+ errorText_ = "RtApiOss::getDeviceInfo: no devices found!";
-+ error( RtAudioError::INVALID_USE );
-+ return info;
-+ }
-+
-+ if ( device >= nDevices ) {
-+ close( mixerfd );
-+ errorText_ = "RtApiOss::getDeviceInfo: device ID is invalid!";
-+ error( RtAudioError::INVALID_USE );
-+ return info;
-+ }
-+
-+ oss_audioinfo ainfo;
-+ ainfo.dev = device;
-+ result = ioctl( mixerfd, SNDCTL_AUDIOINFO, &ainfo );
-+ close( mixerfd );
-+ if ( result == -1 ) {
-+ errorStream_ << "RtApiOss::getDeviceInfo: error getting device (" << ainfo.name << ") info.";
-+ errorText_ = errorStream_.str();
-+ error( RtAudioError::WARNING );
-+ return info;
-+ }
-+
-+ // Probe channels
-+ if ( ainfo.caps & PCM_CAP_OUTPUT ) info.outputChannels = ainfo.max_channels;
-+ if ( ainfo.caps & PCM_CAP_INPUT ) info.inputChannels = ainfo.max_channels;
-+ if ( ainfo.caps & PCM_CAP_DUPLEX ) {
-+ if ( info.outputChannels > 0 && info.inputChannels > 0 && ainfo.caps & PCM_CAP_DUPLEX )
-+ info.duplexChannels = (info.outputChannels > info.inputChannels) ? info.inputChannels : info.outputChannels;
-+ }
-+
-+ // Probe data formats ... do for input
-+ unsigned long mask = ainfo.iformats;
-+ if ( mask & AFMT_S16_LE || mask & AFMT_S16_BE )
-+ info.nativeFormats |= RTAUDIO_SINT16;
-+ if ( mask & AFMT_S8 )
-+ info.nativeFormats |= RTAUDIO_SINT8;
-+ if ( mask & AFMT_S32_LE || mask & AFMT_S32_BE )
-+ info.nativeFormats |= RTAUDIO_SINT32;
-+#ifdef AFMT_FLOAT
-+ if ( mask & AFMT_FLOAT )
-+ info.nativeFormats |= RTAUDIO_FLOAT32;
-+#endif
-+ if ( mask & AFMT_S24_LE || mask & AFMT_S24_BE )
-+ info.nativeFormats |= RTAUDIO_SINT24;
-+
-+ // Check that we have at least one supported format
-+ if ( info.nativeFormats == 0 ) {
-+ errorStream_ << "RtApiOss::getDeviceInfo: device (" << ainfo.name << ") data format not supported by RtAudio.";
-+ errorText_ = errorStream_.str();
-+ error( RtAudioError::WARNING );
-+ return info;
-+ }
-+
-+ // Probe the supported sample rates.
-+ info.sampleRates.clear();
-+ if ( ainfo.nrates ) {
-+ for ( unsigned int i=0; i<ainfo.nrates; i++ ) {
-+ for ( unsigned int k=0; k<MAX_SAMPLE_RATES; k++ ) {
-+ if ( ainfo.rates[i] == SAMPLE_RATES[k] ) {
-+ info.sampleRates.push_back( SAMPLE_RATES[k] );
-+
-+ if ( !info.preferredSampleRate || ( SAMPLE_RATES[k] <= 48000 && SAMPLE_RATES[k] > info.preferredSampleRate ) )
-+ info.preferredSampleRate = SAMPLE_RATES[k];
-+
-+ break;
-+ }
-+ }
-+ }
-+ }
-+ else {
-+ // Check min and max rate values;
-+ for ( unsigned int k=0; k<MAX_SAMPLE_RATES; k++ ) {
-+ if ( ainfo.min_rate <= (int) SAMPLE_RATES[k] && ainfo.max_rate >= (int) SAMPLE_RATES[k] ) {
-+ info.sampleRates.push_back( SAMPLE_RATES[k] );
-+
-+ if ( !info.preferredSampleRate || ( SAMPLE_RATES[k] <= 48000 && SAMPLE_RATES[k] > info.preferredSampleRate ) )
-+ info.preferredSampleRate = SAMPLE_RATES[k];
-+ }
-+ }
-+ }
-+
-+ if ( info.sampleRates.size() == 0 ) {
-+ errorStream_ << "RtApiOss::getDeviceInfo: no supported sample rates found for device (" << ainfo.name << ").";
-+ errorText_ = errorStream_.str();
-+ error( RtAudioError::WARNING );
-+ }
-+ else {
-+ info.probed = true;
-+ info.name = ainfo.name;
-+ }
-+
-+ return info;
-+}
-+
-+
-+bool RtApiOss :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,
-+ unsigned int firstChannel, unsigned int sampleRate,
-+ RtAudioFormat format, unsigned int *bufferSize,
-+ RtAudio::StreamOptions *options )
-+{
-+ int mixerfd = open( "/dev/mixer", O_RDWR, 0 );
-+ if ( mixerfd == -1 ) {
-+ errorText_ = "RtApiOss::probeDeviceOpen: error opening '/dev/mixer'.";
-+ return FAILURE;
-+ }
-+
-+ oss_sysinfo sysinfo;
-+ int result = ioctl( mixerfd, SNDCTL_SYSINFO, &sysinfo );
-+ if ( result == -1 ) {
-+ close( mixerfd );
-+ errorText_ = "RtApiOss::probeDeviceOpen: error getting sysinfo, OSS version >= 4.0 is required.";
-+ return FAILURE;
-+ }
-+
-+ unsigned nDevices = sysinfo.numaudios;
-+ if ( nDevices == 0 ) {
-+ // This should not happen because a check is made before this function is called.
-+ close( mixerfd );
-+ errorText_ = "RtApiOss::probeDeviceOpen: no devices found!";
-+ return FAILURE;
-+ }
-+
-+ if ( device >= nDevices ) {
-+ // This should not happen because a check is made before this function is called.
-+ close( mixerfd );
-+ errorText_ = "RtApiOss::probeDeviceOpen: device ID is invalid!";
-+ return FAILURE;
-+ }
-+
-+ oss_audioinfo ainfo;
-+ ainfo.dev = device;
-+ result = ioctl( mixerfd, SNDCTL_AUDIOINFO, &ainfo );
-+ close( mixerfd );
-+ if ( result == -1 ) {
-+ errorStream_ << "RtApiOss::getDeviceInfo: error getting device (" << ainfo.name << ") info.";
-+ errorText_ = errorStream_.str();
-+ return FAILURE;
-+ }
-+
-+ // Check if device supports input or output
-+ if ( ( mode == OUTPUT && !( ainfo.caps & PCM_CAP_OUTPUT ) ) ||
-+ ( mode == INPUT && !( ainfo.caps & PCM_CAP_INPUT ) ) ) {
-+ if ( mode == OUTPUT )
-+ errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") does not support output.";
-+ else
-+ errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") does not support input.";
-+ errorText_ = errorStream_.str();
-+ return FAILURE;
-+ }
-+
-+ int flags = 0;
-+ OssHandle *handle = (OssHandle *) stream_.apiHandle;
-+ if ( mode == OUTPUT )
-+ flags |= O_WRONLY;
-+ else { // mode == INPUT
-+ if (stream_.mode == OUTPUT && stream_.device[0] == device) {
-+ // We just set the same device for playback ... close and reopen for duplex (OSS only).
-+ close( handle->id[0] );
-+ handle->id[0] = 0;
-+ if ( !( ainfo.caps & PCM_CAP_DUPLEX ) ) {
-+ errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") does not support duplex mode.";
-+ errorText_ = errorStream_.str();
-+ return FAILURE;
-+ }
-+ // Check that the number previously set channels is the same.
-+ if ( stream_.nUserChannels[0] != channels ) {
-+ errorStream_ << "RtApiOss::probeDeviceOpen: input/output channels must be equal for OSS duplex device (" << ainfo.name << ").";
-+ errorText_ = errorStream_.str();
-+ return FAILURE;
-+ }
-+ flags |= O_RDWR;
-+ }
-+ else
-+ flags |= O_RDONLY;
-+ }
-+
-+ // Set exclusive access if specified.
-+ if ( options && options->flags & RTAUDIO_HOG_DEVICE ) flags |= O_EXCL;
-+
-+ // Try to open the device.
-+ int fd;
-+ fd = open( ainfo.devnode, flags, 0 );
-+ if ( fd == -1 ) {
-+ if ( errno == EBUSY )
-+ errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") is busy.";
-+ else
-+ errorStream_ << "RtApiOss::probeDeviceOpen: error opening device (" << ainfo.name << ").";
-+ errorText_ = errorStream_.str();
-+ return FAILURE;
-+ }
-+
-+ // For duplex operation, specifically set this mode (this doesn't seem to work).
-+ /*
-+ if ( flags | O_RDWR ) {
-+ result = ioctl( fd, SNDCTL_DSP_SETDUPLEX, NULL );
-+ if ( result == -1) {
-+ errorStream_ << "RtApiOss::probeDeviceOpen: error setting duplex mode for device (" << ainfo.name << ").";
-+ errorText_ = errorStream_.str();
-+ return FAILURE;
-+ }
-+ }
-+ */
-+
-+ // Check the device channel support.
-+ stream_.nUserChannels[mode] = channels;
-+ if ( ainfo.max_channels < (int)(channels + firstChannel) ) {
-+ close( fd );
-+ errorStream_ << "RtApiOss::probeDeviceOpen: the device (" << ainfo.name << ") does not support requested channel parameters.";
-+ errorText_ = errorStream_.str();
-+ return FAILURE;
-+ }
-+
-+ // Set the number of channels.
-+ int deviceChannels = channels + firstChannel;
-+ result = ioctl( fd, SNDCTL_DSP_CHANNELS, &deviceChannels );
-+ if ( result == -1 || deviceChannels < (int)(channels + firstChannel) ) {
-+ close( fd );
-+ errorStream_ << "RtApiOss::probeDeviceOpen: error setting channel parameters on device (" << ainfo.name << ").";
-+ errorText_ = errorStream_.str();
-+ return FAILURE;
-+ }
-+ stream_.nDeviceChannels[mode] = deviceChannels;
-+
-+ // Get the data format mask
-+ int mask;
-+ result = ioctl( fd, SNDCTL_DSP_GETFMTS, &mask );
-+ if ( result == -1 ) {
-+ close( fd );
-+ errorStream_ << "RtApiOss::probeDeviceOpen: error getting device (" << ainfo.name << ") data formats.";
-+ errorText_ = errorStream_.str();
-+ return FAILURE;
-+ }
-+
-+ // Determine how to set the device format.
-+ stream_.userFormat = format;
-+ int deviceFormat = -1;
-+ stream_.doByteSwap[mode] = false;
-+ if ( format == RTAUDIO_SINT8 ) {
-+ if ( mask & AFMT_S8 ) {
-+ deviceFormat = AFMT_S8;
-+ stream_.deviceFormat[mode] = RTAUDIO_SINT8;
-+ }
-+ }
-+ else if ( format == RTAUDIO_SINT16 ) {
-+ if ( mask & AFMT_S16_NE ) {
-+ deviceFormat = AFMT_S16_NE;
-+ stream_.deviceFormat[mode] = RTAUDIO_SINT16;
-+ }
-+ else if ( mask & AFMT_S16_OE ) {
-+ deviceFormat = AFMT_S16_OE;
-+ stream_.deviceFormat[mode] = RTAUDIO_SINT16;
-+ stream_.doByteSwap[mode] = true;
-+ }
-+ }
-+ else if ( format == RTAUDIO_SINT24 ) {
-+ if ( mask & AFMT_S24_NE ) {
-+ deviceFormat = AFMT_S24_NE;
-+ stream_.deviceFormat[mode] = RTAUDIO_SINT24;
-+ }
-+ else if ( mask & AFMT_S24_OE ) {
-+ deviceFormat = AFMT_S24_OE;
-+ stream_.deviceFormat[mode] = RTAUDIO_SINT24;
-+ stream_.doByteSwap[mode] = true;
-+ }
-+ }
-+ else if ( format == RTAUDIO_SINT32 ) {
-+ if ( mask & AFMT_S32_NE ) {
-+ deviceFormat = AFMT_S32_NE;
-+ stream_.deviceFormat[mode] = RTAUDIO_SINT32;
-+ }
-+ else if ( mask & AFMT_S32_OE ) {
-+ deviceFormat = AFMT_S32_OE;
-+ stream_.deviceFormat[mode] = RTAUDIO_SINT32;
-+ stream_.doByteSwap[mode] = true;
-+ }
-+ }
-+
-+ if ( deviceFormat == -1 ) {
-+ // The user requested format is not natively supported by the device.
-+ if ( mask & AFMT_S16_NE ) {
-+ deviceFormat = AFMT_S16_NE;
-+ stream_.deviceFormat[mode] = RTAUDIO_SINT16;
-+ }
-+ else if ( mask & AFMT_S32_NE ) {
-+ deviceFormat = AFMT_S32_NE;
-+ stream_.deviceFormat[mode] = RTAUDIO_SINT32;
-+ }
-+ else if ( mask & AFMT_S24_NE ) {
-+ deviceFormat = AFMT_S24_NE;
-+ stream_.deviceFormat[mode] = RTAUDIO_SINT24;
-+ }
-+ else if ( mask & AFMT_S16_OE ) {
-+ deviceFormat = AFMT_S16_OE;
-+ stream_.deviceFormat[mode] = RTAUDIO_SINT16;
-+ stream_.doByteSwap[mode] = true;
-+ }
-+ else if ( mask & AFMT_S32_OE ) {
-+ deviceFormat = AFMT_S32_OE;
-+ stream_.deviceFormat[mode] = RTAUDIO_SINT32;
-+ stream_.doByteSwap[mode] = true;
-+ }
-+ else if ( mask & AFMT_S24_OE ) {
-+ deviceFormat = AFMT_S24_OE;
-+ stream_.deviceFormat[mode] = RTAUDIO_SINT24;
-+ stream_.doByteSwap[mode] = true;
-+ }
-+ else if ( mask & AFMT_S8) {
-+ deviceFormat = AFMT_S8;
-+ stream_.deviceFormat[mode] = RTAUDIO_SINT8;
-+ }
-+ }
-+
-+ if ( stream_.deviceFormat[mode] == 0 ) {
-+ // This really shouldn't happen ...
-+ close( fd );
-+ errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") data format not supported by RtAudio.";
-+ errorText_ = errorStream_.str();
-+ return FAILURE;
-+ }
-+
-+ // Set the data format.
-+ int temp = deviceFormat;
-+ result = ioctl( fd, SNDCTL_DSP_SETFMT, &deviceFormat );
-+ if ( result == -1 || deviceFormat != temp ) {
-+ close( fd );
-+ errorStream_ << "RtApiOss::probeDeviceOpen: error setting data format on device (" << ainfo.name << ").";
-+ errorText_ = errorStream_.str();
-+ return FAILURE;
-+ }
-+
-+ // Attempt to set the buffer size. According to OSS, the minimum
-+ // number of buffers is two. The supposed minimum buffer size is 16
-+ // bytes, so that will be our lower bound. The argument to this
-+ // call is in the form 0xMMMMSSSS (hex), where the buffer size (in
-+ // bytes) is given as 2^SSSS and the number of buffers as 2^MMMM.
-+ // We'll check the actual value used near the end of the setup
-+ // procedure.
-+ int ossBufferBytes = *bufferSize * formatBytes( stream_.deviceFormat[mode] ) * deviceChannels;
-+ if ( ossBufferBytes < 16 ) ossBufferBytes = 16;
-+ int buffers = 0;
-+ if ( options ) buffers = options->numberOfBuffers;
-+ if ( options && options->flags & RTAUDIO_MINIMIZE_LATENCY ) buffers = 2;
-+ if ( buffers < 2 ) buffers = 3;
-+ temp = ((int) buffers << 16) + (int)( log10( (double)ossBufferBytes ) / log10( 2.0 ) );
-+ result = ioctl( fd, SNDCTL_DSP_SETFRAGMENT, &temp );
-+ if ( result == -1 ) {
-+ close( fd );
-+ errorStream_ << "RtApiOss::probeDeviceOpen: error setting buffer size on device (" << ainfo.name << ").";
-+ errorText_ = errorStream_.str();
-+ return FAILURE;
-+ }
-+ stream_.nBuffers = buffers;
-+
-+ // Save buffer size (in sample frames).
-+ *bufferSize = ossBufferBytes / ( formatBytes(stream_.deviceFormat[mode]) * deviceChannels );
-+ stream_.bufferSize = *bufferSize;
-+
-+ // Set the sample rate.
-+ int srate = sampleRate;
-+ result = ioctl( fd, SNDCTL_DSP_SPEED, &srate );
-+ if ( result == -1 ) {
-+ close( fd );
-+ errorStream_ << "RtApiOss::probeDeviceOpen: error setting sample rate (" << sampleRate << ") on device (" << ainfo.name << ").";
-+ errorText_ = errorStream_.str();
-+ return FAILURE;
-+ }
-+
-+ // Verify the sample rate setup worked.
-+ if ( abs( srate - (int)sampleRate ) > 100 ) {
-+ close( fd );
-+ errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") does not support sample rate (" << sampleRate << ").";
-+ errorText_ = errorStream_.str();
-+ return FAILURE;
-+ }
-+ stream_.sampleRate = sampleRate;
-+
-+ if ( mode == INPUT && stream_.mode == OUTPUT && stream_.device[0] == device) {
-+ // We're doing duplex setup here.
-+ stream_.deviceFormat[0] = stream_.deviceFormat[1];
-+ stream_.nDeviceChannels[0] = deviceChannels;
-+ }
-+
-+ // Set interleaving parameters.
-+ stream_.userInterleaved = true;
-+ stream_.deviceInterleaved[mode] = true;
-+ if ( options && options->flags & RTAUDIO_NONINTERLEAVED )
-+ stream_.userInterleaved = false;
-+
-+ // Set flags for buffer conversion
-+ stream_.doConvertBuffer[mode] = false;
-+ if ( stream_.userFormat != stream_.deviceFormat[mode] )
-+ stream_.doConvertBuffer[mode] = true;
-+ if ( stream_.nUserChannels[mode] < stream_.nDeviceChannels[mode] )
-+ stream_.doConvertBuffer[mode] = true;
-+ if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] &&
-+ stream_.nUserChannels[mode] > 1 )
-+ stream_.doConvertBuffer[mode] = true;
-+
-+ // Allocate the stream handles if necessary and then save.
-+ if ( stream_.apiHandle == 0 ) {
-+ try {
-+ handle = new OssHandle;
-+ }
-+ catch ( std::bad_alloc& ) {
-+ errorText_ = "RtApiOss::probeDeviceOpen: error allocating OssHandle memory.";
-+ goto error;
-+ }
-+
-+ if ( pthread_cond_init( &handle->runnable, NULL ) ) {
-+ errorText_ = "RtApiOss::probeDeviceOpen: error initializing pthread condition variable.";
-+ goto error;
-+ }
-+
-+ stream_.apiHandle = (void *) handle;
-+ }
-+ else {
-+ handle = (OssHandle *) stream_.apiHandle;
-+ }
-+ handle->id[mode] = fd;
-+
-+ // Allocate necessary internal buffers.
-+ unsigned long bufferBytes;
-+ bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat );
-+ stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 );
-+ if ( stream_.userBuffer[mode] == NULL ) {
-+ errorText_ = "RtApiOss::probeDeviceOpen: error allocating user buffer memory.";
-+ goto error;
-+ }
-+
-+ if ( stream_.doConvertBuffer[mode] ) {
-+
-+ bool makeBuffer = true;
-+ bufferBytes = stream_.nDeviceChannels[mode] * formatBytes( stream_.deviceFormat[mode] );
-+ if ( mode == INPUT ) {
-+ if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) {
-+ unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] );
-+ if ( bufferBytes <= bytesOut ) makeBuffer = false;
-+ }
-+ }
-+
-+ if ( makeBuffer ) {
-+ bufferBytes *= *bufferSize;
-+ if ( stream_.deviceBuffer ) free( stream_.deviceBuffer );
-+ stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 );
-+ if ( stream_.deviceBuffer == NULL ) {
-+ errorText_ = "RtApiOss::probeDeviceOpen: error allocating device buffer memory.";
-+ goto error;
-+ }
-+ }
-+ }
-+
-+ stream_.device[mode] = device;
-+ stream_.state = STREAM_STOPPED;
-+
-+ // Setup the buffer conversion information structure.
-+ if ( stream_.doConvertBuffer[mode] ) setConvertInfo( mode, firstChannel );
-+
-+ // Setup thread if necessary.
-+ if ( stream_.mode == OUTPUT && mode == INPUT ) {
-+ // We had already set up an output stream.
-+ stream_.mode = DUPLEX;
-+ if ( stream_.device[0] == device ) handle->id[0] = fd;
-+ }
-+ else {
-+ stream_.mode = mode;
-+
-+ // Setup callback thread.
-+ stream_.callbackInfo.object = (void *) this;
-+
-+ // Set the thread attributes for joinable and realtime scheduling
-+ // priority. The higher priority will only take affect if the
-+ // program is run as root or suid.
-+ pthread_attr_t attr;
-+ pthread_attr_init( &attr );
-+ pthread_attr_setdetachstate( &attr, PTHREAD_CREATE_JOINABLE );
-+#ifdef SCHED_RR // Undefined with some OSes (eg: NetBSD 1.6.x with GNU Pthread)
-+ if ( options && options->flags & RTAUDIO_SCHEDULE_REALTIME ) {
-+ struct sched_param param;
-+ int priority = options->priority;
-+ int min = sched_get_priority_min( SCHED_RR );
-+ int max = sched_get_priority_max( SCHED_RR );
-+ if ( priority < min ) priority = min;
-+ else if ( priority > max ) priority = max;
-+ param.sched_priority = priority;
-+ pthread_attr_setschedparam( &attr, ¶m );
-+ pthread_attr_setschedpolicy( &attr, SCHED_RR );
-+ }
-+ else
-+ pthread_attr_setschedpolicy( &attr, SCHED_OTHER );
-+#else
-+ pthread_attr_setschedpolicy( &attr, SCHED_OTHER );
-+#endif
-+
-+ stream_.callbackInfo.isRunning = true;
-+ result = pthread_create( &stream_.callbackInfo.thread, &attr, ossCallbackHandler, &stream_.callbackInfo );
-+ pthread_attr_destroy( &attr );
-+ if ( result ) {
-+ stream_.callbackInfo.isRunning = false;
-+ errorText_ = "RtApiOss::error creating callback thread!";
-+ goto error;
-+ }
-+ }
-+
-+ return SUCCESS;
-+
-+ error:
-+ if ( handle ) {
-+ pthread_cond_destroy( &handle->runnable );
-+ if ( handle->id[0] ) close( handle->id[0] );
-+ if ( handle->id[1] ) close( handle->id[1] );
-+ delete handle;
-+ stream_.apiHandle = 0;
-+ }
-+
-+ for ( int i=0; i<2; i++ ) {
-+ if ( stream_.userBuffer[i] ) {
-+ free( stream_.userBuffer[i] );
-+ stream_.userBuffer[i] = 0;
-+ }
-+ }
-+
-+ if ( stream_.deviceBuffer ) {
-+ free( stream_.deviceBuffer );
-+ stream_.deviceBuffer = 0;
-+ }
-+
-+ return FAILURE;
-+}
-+
-+void RtApiOss :: closeStream()
-+{
-+ if ( stream_.state == STREAM_CLOSED ) {
-+ errorText_ = "RtApiOss::closeStream(): no open stream to close!";
-+ error( RtAudioError::WARNING );
-+ return;
-+ }
-+
-+ OssHandle *handle = (OssHandle *) stream_.apiHandle;
-+ stream_.callbackInfo.isRunning = false;
-+ MUTEX_LOCK( &stream_.mutex );
-+ if ( stream_.state == STREAM_STOPPED )
-+ pthread_cond_signal( &handle->runnable );
-+ MUTEX_UNLOCK( &stream_.mutex );
-+ pthread_join( stream_.callbackInfo.thread, NULL );
-+
-+ if ( stream_.state == STREAM_RUNNING ) {
-+ if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX )
-+ ioctl( handle->id[0], SNDCTL_DSP_HALT, 0 );
-+ else
-+ ioctl( handle->id[1], SNDCTL_DSP_HALT, 0 );
-+ stream_.state = STREAM_STOPPED;
-+ }
-+
-+ if ( handle ) {
-+ pthread_cond_destroy( &handle->runnable );
-+ if ( handle->id[0] ) close( handle->id[0] );
-+ if ( handle->id[1] ) close( handle->id[1] );
-+ delete handle;
-+ stream_.apiHandle = 0;
-+ }
-+
-+ for ( int i=0; i<2; i++ ) {
-+ if ( stream_.userBuffer[i] ) {
-+ free( stream_.userBuffer[i] );
-+ stream_.userBuffer[i] = 0;
-+ }
-+ }
-+
-+ if ( stream_.deviceBuffer ) {
-+ free( stream_.deviceBuffer );
-+ stream_.deviceBuffer = 0;
-+ }
-+
-+ stream_.mode = UNINITIALIZED;
-+ stream_.state = STREAM_CLOSED;
-+}
-+
-+void RtApiOss :: startStream()
-+{
-+ verifyStream();
-+ if ( stream_.state == STREAM_RUNNING ) {
-+ errorText_ = "RtApiOss::startStream(): the stream is already running!";
-+ error( RtAudioError::WARNING );
-+ return;
-+ }
-+
-+ MUTEX_LOCK( &stream_.mutex );
-+
-+ stream_.state = STREAM_RUNNING;
-+
-+ // No need to do anything else here ... OSS automatically starts
-+ // when fed samples.
-+
-+ MUTEX_UNLOCK( &stream_.mutex );
-+
-+ OssHandle *handle = (OssHandle *) stream_.apiHandle;
-+ pthread_cond_signal( &handle->runnable );
-+}
-+
-+void RtApiOss :: stopStream()
-+{
-+ verifyStream();
-+ if ( stream_.state == STREAM_STOPPED ) {
-+ errorText_ = "RtApiOss::stopStream(): the stream is already stopped!";
-+ error( RtAudioError::WARNING );
-+ return;
-+ }
-+
-+ MUTEX_LOCK( &stream_.mutex );
-+
-+ // The state might change while waiting on a mutex.
-+ if ( stream_.state == STREAM_STOPPED ) {
-+ MUTEX_UNLOCK( &stream_.mutex );
-+ return;
-+ }
-+
-+ int result = 0;
-+ OssHandle *handle = (OssHandle *) stream_.apiHandle;
-+ if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
-+
-+ // Flush the output with zeros a few times.
-+ char *buffer;
-+ int samples;
-+ RtAudioFormat format;
-+
-+ if ( stream_.doConvertBuffer[0] ) {
-+ buffer = stream_.deviceBuffer;
-+ samples = stream_.bufferSize * stream_.nDeviceChannels[0];
-+ format = stream_.deviceFormat[0];
-+ }
-+ else {
-+ buffer = stream_.userBuffer[0];
-+ samples = stream_.bufferSize * stream_.nUserChannels[0];
-+ format = stream_.userFormat;
-+ }
-+
-+ memset( buffer, 0, samples * formatBytes(format) );
-+ for ( unsigned int i=0; i<stream_.nBuffers+1; i++ ) {
-+ result = write( handle->id[0], buffer, samples * formatBytes(format) );
-+ if ( result == -1 ) {
-+ errorText_ = "RtApiOss::stopStream: audio write error.";
-+ error( RtAudioError::WARNING );
-+ }
-+ }
-+
-+ result = ioctl( handle->id[0], SNDCTL_DSP_HALT, 0 );
-+ if ( result == -1 ) {
-+ errorStream_ << "RtApiOss::stopStream: system error stopping callback procedure on device (" << stream_.device[0] << ").";
-+ errorText_ = errorStream_.str();
-+ goto unlock;
-+ }
-+ handle->triggered = false;
-+ }
-+
-+ if ( stream_.mode == INPUT || ( stream_.mode == DUPLEX && handle->id[0] != handle->id[1] ) ) {
-+ result = ioctl( handle->id[1], SNDCTL_DSP_HALT, 0 );
-+ if ( result == -1 ) {
-+ errorStream_ << "RtApiOss::stopStream: system error stopping input callback procedure on device (" << stream_.device[0] << ").";
-+ errorText_ = errorStream_.str();
-+ goto unlock;
-+ }
-+ }
-+
-+ unlock:
-+ stream_.state = STREAM_STOPPED;
-+ MUTEX_UNLOCK( &stream_.mutex );
-+
-+ if ( result != -1 ) return;
-+ error( RtAudioError::SYSTEM_ERROR );
-+}
-+
-+void RtApiOss :: abortStream()
-+{
-+ verifyStream();
-+ if ( stream_.state == STREAM_STOPPED ) {
-+ errorText_ = "RtApiOss::abortStream(): the stream is already stopped!";
-+ error( RtAudioError::WARNING );
-+ return;
-+ }
-+
-+ MUTEX_LOCK( &stream_.mutex );
-+
-+ // The state might change while waiting on a mutex.
-+ if ( stream_.state == STREAM_STOPPED ) {
-+ MUTEX_UNLOCK( &stream_.mutex );
-+ return;
-+ }
-+
-+ int result = 0;
-+ OssHandle *handle = (OssHandle *) stream_.apiHandle;
-+ if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
-+ result = ioctl( handle->id[0], SNDCTL_DSP_HALT, 0 );
-+ if ( result == -1 ) {
-+ errorStream_ << "RtApiOss::abortStream: system error stopping callback procedure on device (" << stream_.device[0] << ").";
-+ errorText_ = errorStream_.str();
-+ goto unlock;
-+ }
-+ handle->triggered = false;
-+ }
-+
-+ if ( stream_.mode == INPUT || ( stream_.mode == DUPLEX && handle->id[0] != handle->id[1] ) ) {
-+ result = ioctl( handle->id[1], SNDCTL_DSP_HALT, 0 );
-+ if ( result == -1 ) {
-+ errorStream_ << "RtApiOss::abortStream: system error stopping input callback procedure on device (" << stream_.device[0] << ").";
-+ errorText_ = errorStream_.str();
-+ goto unlock;
-+ }
-+ }
-+
-+ unlock:
-+ stream_.state = STREAM_STOPPED;
-+ MUTEX_UNLOCK( &stream_.mutex );
-+
-+ if ( result != -1 ) return;
-+ error( RtAudioError::SYSTEM_ERROR );
-+}
-+
-+void RtApiOss :: callbackEvent()
-+{
-+ OssHandle *handle = (OssHandle *) stream_.apiHandle;
-+ if ( stream_.state == STREAM_STOPPED ) {
-+ MUTEX_LOCK( &stream_.mutex );
-+ pthread_cond_wait( &handle->runnable, &stream_.mutex );
-+ if ( stream_.state != STREAM_RUNNING ) {
-+ MUTEX_UNLOCK( &stream_.mutex );
-+ return;
-+ }
-+ MUTEX_UNLOCK( &stream_.mutex );
-+ }
-+
-+ if ( stream_.state == STREAM_CLOSED ) {
-+ errorText_ = "RtApiOss::callbackEvent(): the stream is closed ... this shouldn't happen!";
-+ error( RtAudioError::WARNING );
-+ return;
-+ }
-+
-+ // Invoke user callback to get fresh output data.
-+ int doStopStream = 0;
-+ RtAudioCallback callback = (RtAudioCallback) stream_.callbackInfo.callback;
-+ double streamTime = getStreamTime();
-+ RtAudioStreamStatus status = 0;
-+ if ( stream_.mode != INPUT && handle->xrun[0] == true ) {
-+ status |= RTAUDIO_OUTPUT_UNDERFLOW;
-+ handle->xrun[0] = false;
-+ }
-+ if ( stream_.mode != OUTPUT && handle->xrun[1] == true ) {
-+ status |= RTAUDIO_INPUT_OVERFLOW;
-+ handle->xrun[1] = false;
-+ }
-+ doStopStream = callback( stream_.userBuffer[0], stream_.userBuffer[1],
-+ stream_.bufferSize, streamTime, status, stream_.callbackInfo.userData );
-+ if ( doStopStream == 2 ) {
-+ this->abortStream();
-+ return;
-+ }
-+
-+ MUTEX_LOCK( &stream_.mutex );
-+
-+ // The state might change while waiting on a mutex.
-+ if ( stream_.state == STREAM_STOPPED ) goto unlock;
-+
-+ int result;
-+ char *buffer;
-+ int samples;
-+ RtAudioFormat format;
-+
-+ if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
-+
-+ // Setup parameters and do buffer conversion if necessary.
-+ if ( stream_.doConvertBuffer[0] ) {
-+ buffer = stream_.deviceBuffer;
-+ convertBuffer( buffer, stream_.userBuffer[0], stream_.convertInfo[0] );
-+ samples = stream_.bufferSize * stream_.nDeviceChannels[0];
-+ format = stream_.deviceFormat[0];
-+ }
-+ else {
-+ buffer = stream_.userBuffer[0];
-+ samples = stream_.bufferSize * stream_.nUserChannels[0];
-+ format = stream_.userFormat;
-+ }
-+
-+ // Do byte swapping if necessary.
-+ if ( stream_.doByteSwap[0] )
-+ byteSwapBuffer( buffer, samples, format );
-+
-+ if ( stream_.mode == DUPLEX && handle->triggered == false ) {
-+ int trig = 0;
-+ ioctl( handle->id[0], SNDCTL_DSP_SETTRIGGER, &trig );
-+ result = write( handle->id[0], buffer, samples * formatBytes(format) );
-+ trig = PCM_ENABLE_INPUT|PCM_ENABLE_OUTPUT;
-+ ioctl( handle->id[0], SNDCTL_DSP_SETTRIGGER, &trig );
-+ handle->triggered = true;
-+ }
-+ else
-+ // Write samples to device.
-+ result = write( handle->id[0], buffer, samples * formatBytes(format) );
-+
-+ if ( result == -1 ) {
-+ // We'll assume this is an underrun, though there isn't a
-+ // specific means for determining that.
-+ handle->xrun[0] = true;
-+ errorText_ = "RtApiOss::callbackEvent: audio write error.";
-+ error( RtAudioError::WARNING );
-+ // Continue on to input section.
-+ }
-+ }
-+
-+ if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {
-+
-+ // Setup parameters.
-+ if ( stream_.doConvertBuffer[1] ) {
-+ buffer = stream_.deviceBuffer;
-+ samples = stream_.bufferSize * stream_.nDeviceChannels[1];
-+ format = stream_.deviceFormat[1];
-+ }
-+ else {
-+ buffer = stream_.userBuffer[1];
-+ samples = stream_.bufferSize * stream_.nUserChannels[1];
-+ format = stream_.userFormat;
-+ }
-+
-+ // Read samples from device.
-+ result = read( handle->id[1], buffer, samples * formatBytes(format) );
-+
-+ if ( result == -1 ) {
-+ // We'll assume this is an overrun, though there isn't a
-+ // specific means for determining that.
-+ handle->xrun[1] = true;
-+ errorText_ = "RtApiOss::callbackEvent: audio read error.";
-+ error( RtAudioError::WARNING );
-+ goto unlock;
-+ }
-+
-+ // Do byte swapping if necessary.
-+ if ( stream_.doByteSwap[1] )
-+ byteSwapBuffer( buffer, samples, format );
-+
-+ // Do buffer conversion if necessary.
-+ if ( stream_.doConvertBuffer[1] )
-+ convertBuffer( stream_.userBuffer[1], stream_.deviceBuffer, stream_.convertInfo[1] );
-+ }
-+
-+ unlock:
-+ MUTEX_UNLOCK( &stream_.mutex );
-+
-+ RtApi::tickStreamTime();
-+ if ( doStopStream == 1 ) this->stopStream();
-+}
-+
-+static void *ossCallbackHandler( void *ptr )
-+{
-+ CallbackInfo *info = (CallbackInfo *) ptr;
-+ RtApiOss *object = (RtApiOss *) info->object;
-+ bool *isRunning = &info->isRunning;
-+
-+ while ( *isRunning == true ) {
-+ pthread_testcancel();
-+ object->callbackEvent();
-+ }
-+
-+ pthread_exit( NULL );
-+}
-+
-+//******************** End of __LINUX_OSS__ *********************//
-+#endif
-+
-+
-+// *************************************************** //
-+//
-+// Protected common (OS-independent) RtAudio methods.
-+//
-+// *************************************************** //
-+
-+// This method can be modified to control the behavior of error
-+// message printing.
-+void RtApi :: error( RtAudioError::Type type )
-+{
-+ errorStream_.str(""); // clear the ostringstream
-+
-+ RtAudioErrorCallback errorCallback = (RtAudioErrorCallback) stream_.callbackInfo.errorCallback;
-+ if ( errorCallback ) {
-+ // abortStream() can generate new error messages. Ignore them. Just keep original one.
-+
-+ if ( firstErrorOccurred_ )
-+ return;
-+
-+ firstErrorOccurred_ = true;
-+ const std::string errorMessage = errorText_;
-+
-+ if ( type != RtAudioError::WARNING && stream_.state != STREAM_STOPPED) {
-+ stream_.callbackInfo.isRunning = false; // exit from the thread
-+ abortStream();
-+ }
-+
-+ errorCallback( type, errorMessage );
-+ firstErrorOccurred_ = false;
-+ return;
-+ }
-+
-+ if ( type == RtAudioError::WARNING && showWarnings_ == true )
-+ std::cerr << '\n' << errorText_ << "\n\n";
-+ else if ( type != RtAudioError::WARNING )
-+ throw( RtAudioError( errorText_, type ) );
-+}
-+
-+void RtApi :: verifyStream()
-+{
-+ if ( stream_.state == STREAM_CLOSED ) {
-+ errorText_ = "RtApi:: a stream is not open!";
-+ error( RtAudioError::INVALID_USE );
-+ }
-+}
-+
-+void RtApi :: clearStreamInfo()
-+{
-+ stream_.mode = UNINITIALIZED;
-+ stream_.state = STREAM_CLOSED;
-+ stream_.sampleRate = 0;
-+ stream_.bufferSize = 0;
-+ stream_.nBuffers = 0;
-+ stream_.userFormat = 0;
-+ stream_.userInterleaved = true;
-+ stream_.streamTime = 0.0;
-+ stream_.apiHandle = 0;
-+ stream_.deviceBuffer = 0;
-+ stream_.callbackInfo.callback = 0;
-+ stream_.callbackInfo.userData = 0;
-+ stream_.callbackInfo.isRunning = false;
-+ stream_.callbackInfo.errorCallback = 0;
-+ for ( int i=0; i<2; i++ ) {
-+ stream_.device[i] = 11111;
-+ stream_.doConvertBuffer[i] = false;
-+ stream_.deviceInterleaved[i] = true;
-+ stream_.doByteSwap[i] = false;
-+ stream_.nUserChannels[i] = 0;
-+ stream_.nDeviceChannels[i] = 0;
-+ stream_.channelOffset[i] = 0;
-+ stream_.deviceFormat[i] = 0;
-+ stream_.latency[i] = 0;
-+ stream_.userBuffer[i] = 0;
-+ stream_.convertInfo[i].channels = 0;
-+ stream_.convertInfo[i].inJump = 0;
-+ stream_.convertInfo[i].outJump = 0;
-+ stream_.convertInfo[i].inFormat = 0;
-+ stream_.convertInfo[i].outFormat = 0;
-+ stream_.convertInfo[i].inOffset.clear();
-+ stream_.convertInfo[i].outOffset.clear();
-+ }
-+}
-+
-+unsigned int RtApi :: formatBytes( RtAudioFormat format )
-+{
-+ if ( format == RTAUDIO_SINT16 )
-+ return 2;
-+ else if ( format == RTAUDIO_SINT32 || format == RTAUDIO_FLOAT32 )
-+ return 4;
-+ else if ( format == RTAUDIO_FLOAT64 )
-+ return 8;
-+ else if ( format == RTAUDIO_SINT24 )
-+ return 3;
-+ else if ( format == RTAUDIO_SINT8 )
-+ return 1;
-+
-+ errorText_ = "RtApi::formatBytes: undefined format.";
-+ error( RtAudioError::WARNING );
-+
-+ return 0;
-+}
-+
-+void RtApi :: setConvertInfo( StreamMode mode, unsigned int firstChannel )
-+{
-+ if ( mode == INPUT ) { // convert device to user buffer
-+ stream_.convertInfo[mode].inJump = stream_.nDeviceChannels[1];
-+ stream_.convertInfo[mode].outJump = stream_.nUserChannels[1];
-+ stream_.convertInfo[mode].inFormat = stream_.deviceFormat[1];
-+ stream_.convertInfo[mode].outFormat = stream_.userFormat;
-+ }
-+ else { // convert user to device buffer
-+ stream_.convertInfo[mode].inJump = stream_.nUserChannels[0];
-+ stream_.convertInfo[mode].outJump = stream_.nDeviceChannels[0];
-+ stream_.convertInfo[mode].inFormat = stream_.userFormat;
-+ stream_.convertInfo[mode].outFormat = stream_.deviceFormat[0];
-+ }
-+
-+ if ( stream_.convertInfo[mode].inJump < stream_.convertInfo[mode].outJump )
-+ stream_.convertInfo[mode].channels = stream_.convertInfo[mode].inJump;
-+ else
-+ stream_.convertInfo[mode].channels = stream_.convertInfo[mode].outJump;
-+
-+ // Set up the interleave/deinterleave offsets.
-+ if ( stream_.deviceInterleaved[mode] != stream_.userInterleaved ) {
-+ if ( ( mode == OUTPUT && stream_.deviceInterleaved[mode] ) ||
-+ ( mode == INPUT && stream_.userInterleaved ) ) {
-+ for ( int k=0; k<stream_.convertInfo[mode].channels; k++ ) {
-+ stream_.convertInfo[mode].inOffset.push_back( k * stream_.bufferSize );
-+ stream_.convertInfo[mode].outOffset.push_back( k );
-+ stream_.convertInfo[mode].inJump = 1;
-+ }
-+ }
-+ else {
-+ for ( int k=0; k<stream_.convertInfo[mode].channels; k++ ) {
-+ stream_.convertInfo[mode].inOffset.push_back( k );
-+ stream_.convertInfo[mode].outOffset.push_back( k * stream_.bufferSize );
-+ stream_.convertInfo[mode].outJump = 1;
-+ }
-+ }
-+ }
-+ else { // no (de)interleaving
-+ if ( stream_.userInterleaved ) {
-+ for ( int k=0; k<stream_.convertInfo[mode].channels; k++ ) {
-+ stream_.convertInfo[mode].inOffset.push_back( k );
-+ stream_.convertInfo[mode].outOffset.push_back( k );
-+ }
-+ }
-+ else {
-+ for ( int k=0; k<stream_.convertInfo[mode].channels; k++ ) {
-+ stream_.convertInfo[mode].inOffset.push_back( k * stream_.bufferSize );
-+ stream_.convertInfo[mode].outOffset.push_back( k * stream_.bufferSize );
-+ stream_.convertInfo[mode].inJump = 1;
-+ stream_.convertInfo[mode].outJump = 1;
-+ }
-+ }
-+ }
-+
-+ // Add channel offset.
-+ if ( firstChannel > 0 ) {
-+ if ( stream_.deviceInterleaved[mode] ) {
-+ if ( mode == OUTPUT ) {
-+ for ( int k=0; k<stream_.convertInfo[mode].channels; k++ )
-+ stream_.convertInfo[mode].outOffset[k] += firstChannel;
-+ }
-+ else {
-+ for ( int k=0; k<stream_.convertInfo[mode].channels; k++ )
-+ stream_.convertInfo[mode].inOffset[k] += firstChannel;
-+ }
-+ }
-+ else {
-+ if ( mode == OUTPUT ) {
-+ for ( int k=0; k<stream_.convertInfo[mode].channels; k++ )
-+ stream_.convertInfo[mode].outOffset[k] += ( firstChannel * stream_.bufferSize );
-+ }
-+ else {
-+ for ( int k=0; k<stream_.convertInfo[mode].channels; k++ )
-+ stream_.convertInfo[mode].inOffset[k] += ( firstChannel * stream_.bufferSize );
-+ }
-+ }
-+ }
-+}
-+
-+void RtApi :: convertBuffer( char *outBuffer, char *inBuffer, ConvertInfo &info )
-+{
-+ // This function does format conversion, input/output channel compensation, and
-+ // data interleaving/deinterleaving. 24-bit integers are assumed to occupy
-+ // the lower three bytes of a 32-bit integer.
-+
-+ // Clear our device buffer when in/out duplex device channels are different
-+ if ( outBuffer == stream_.deviceBuffer && stream_.mode == DUPLEX &&
-+ ( stream_.nDeviceChannels[0] < stream_.nDeviceChannels[1] ) )
-+ memset( outBuffer, 0, stream_.bufferSize * info.outJump * formatBytes( info.outFormat ) );
-+
-+ int j;
-+ if (info.outFormat == RTAUDIO_FLOAT64) {
-+ Float64 scale;
-+ Float64 *out = (Float64 *)outBuffer;
-+
-+ if (info.inFormat == RTAUDIO_SINT8) {
-+ signed char *in = (signed char *)inBuffer;
-+ scale = 1.0 / 127.5;
-+ for (unsigned int i=0; i<stream_.bufferSize; i++) {
-+ for (j=0; j<info.channels; j++) {
-+ out[info.outOffset[j]] = (Float64) in[info.inOffset[j]];
-+ out[info.outOffset[j]] += 0.5;
-+ out[info.outOffset[j]] *= scale;
-+ }
-+ in += info.inJump;
-+ out += info.outJump;
-+ }
-+ }
-+ else if (info.inFormat == RTAUDIO_SINT16) {
-+ Int16 *in = (Int16 *)inBuffer;
-+ scale = 1.0 / 32767.5;
-+ for (unsigned int i=0; i<stream_.bufferSize; i++) {
-+ for (j=0; j<info.channels; j++) {
-+ out[info.outOffset[j]] = (Float64) in[info.inOffset[j]];
-+ out[info.outOffset[j]] += 0.5;
-+ out[info.outOffset[j]] *= scale;
-+ }
-+ in += info.inJump;
-+ out += info.outJump;
-+ }
-+ }
-+ else if (info.inFormat == RTAUDIO_SINT24) {
-+ Int24 *in = (Int24 *)inBuffer;
-+ scale = 1.0 / 8388607.5;
-+ for (unsigned int i=0; i<stream_.bufferSize; i++) {
-+ for (j=0; j<info.channels; j++) {
-+ out[info.outOffset[j]] = (Float64) (in[info.inOffset[j]].asInt());
-+ out[info.outOffset[j]] += 0.5;
-+ out[info.outOffset[j]] *= scale;
-+ }
-+ in += info.inJump;
-+ out += info.outJump;
-+ }
-+ }
-+ else if (info.inFormat == RTAUDIO_SINT32) {
-+ Int32 *in = (Int32 *)inBuffer;
-+ scale = 1.0 / 2147483647.5;
-+ for (unsigned int i=0; i<stream_.bufferSize; i++) {
-+ for (j=0; j<info.channels; j++) {
-+ out[info.outOffset[j]] = (Float64) in[info.inOffset[j]];
-+ out[info.outOffset[j]] += 0.5;
-+ out[info.outOffset[j]] *= scale;
-+ }
-+ in += info.inJump;
-+ out += info.outJump;
-+ }
-+ }
-+ else if (info.inFormat == RTAUDIO_FLOAT32) {
-+ Float32 *in = (Float32 *)inBuffer;
-+ for (unsigned int i=0; i<stream_.bufferSize; i++) {
-+ for (j=0; j<info.channels; j++) {
-+ out[info.outOffset[j]] = (Float64) in[info.inOffset[j]];
-+ }
-+ in += info.inJump;
-+ out += info.outJump;
-+ }
-+ }
-+ else if (info.inFormat == RTAUDIO_FLOAT64) {
-+ // Channel compensation and/or (de)interleaving only.
-+ Float64 *in = (Float64 *)inBuffer;
-+ for (unsigned int i=0; i<stream_.bufferSize; i++) {
-+ for (j=0; j<info.channels; j++) {
-+ out[info.outOffset[j]] = in[info.inOffset[j]];
-+ }
-+ in += info.inJump;
-+ out += info.outJump;
-+ }
-+ }
-+ }
-+ else if (info.outFormat == RTAUDIO_FLOAT32) {
-+ Float32 scale;
-+ Float32 *out = (Float32 *)outBuffer;
-+
-+ if (info.inFormat == RTAUDIO_SINT8) {
-+ signed char *in = (signed char *)inBuffer;
-+ scale = (Float32) ( 1.0 / 127.5 );
-+ for (unsigned int i=0; i<stream_.bufferSize; i++) {
-+ for (j=0; j<info.channels; j++) {
-+ out[info.outOffset[j]] = (Float32) in[info.inOffset[j]];
-+ out[info.outOffset[j]] += 0.5;
-+ out[info.outOffset[j]] *= scale;
-+ }
-+ in += info.inJump;
-+ out += info.outJump;
-+ }
-+ }
-+ else if (info.inFormat == RTAUDIO_SINT16) {
-+ Int16 *in = (Int16 *)inBuffer;
-+ scale = (Float32) ( 1.0 / 32767.5 );
-+ for (unsigned int i=0; i<stream_.bufferSize; i++) {
-+ for (j=0; j<info.channels; j++) {
-+ out[info.outOffset[j]] = (Float32) in[info.inOffset[j]];
-+ out[info.outOffset[j]] += 0.5;
-+ out[info.outOffset[j]] *= scale;
-+ }
-+ in += info.inJump;
-+ out += info.outJump;
-+ }
-+ }
-+ else if (info.inFormat == RTAUDIO_SINT24) {
-+ Int24 *in = (Int24 *)inBuffer;
-+ scale = (Float32) ( 1.0 / 8388607.5 );
-+ for (unsigned int i=0; i<stream_.bufferSize; i++) {
-+ for (j=0; j<info.channels; j++) {
-+ out[info.outOffset[j]] = (Float32) (in[info.inOffset[j]].asInt());
-+ out[info.outOffset[j]] += 0.5;
-+ out[info.outOffset[j]] *= scale;
-+ }
-+ in += info.inJump;
-+ out += info.outJump;
-+ }
-+ }
-+ else if (info.inFormat == RTAUDIO_SINT32) {
-+ Int32 *in = (Int32 *)inBuffer;
-+ scale = (Float32) ( 1.0 / 2147483647.5 );
-+ for (unsigned int i=0; i<stream_.bufferSize; i++) {
-+ for (j=0; j<info.channels; j++) {
-+ out[info.outOffset[j]] = (Float32) in[info.inOffset[j]];
-+ out[info.outOffset[j]] += 0.5;
-+ out[info.outOffset[j]] *= scale;
-+ }
-+ in += info.inJump;
-+ out += info.outJump;
-+ }
-+ }
-+ else if (info.inFormat == RTAUDIO_FLOAT32) {
-+ // Channel compensation and/or (de)interleaving only.
-+ Float32 *in = (Float32 *)inBuffer;
-+ for (unsigned int i=0; i<stream_.bufferSize; i++) {
-+ for (j=0; j<info.channels; j++) {
-+ out[info.outOffset[j]] = in[info.inOffset[j]];
-+ }
-+ in += info.inJump;
-+ out += info.outJump;
-+ }
-+ }
-+ else if (info.inFormat == RTAUDIO_FLOAT64) {
-+ Float64 *in = (Float64 *)inBuffer;
-+ for (unsigned int i=0; i<stream_.bufferSize; i++) {
-+ for (j=0; j<info.channels; j++) {
-+ out[info.outOffset[j]] = (Float32) in[info.inOffset[j]];
-+ }
-+ in += info.inJump;
-+ out += info.outJump;
-+ }
-+ }
-+ }
-+ else if (info.outFormat == RTAUDIO_SINT32) {
-+ Int32 *out = (Int32 *)outBuffer;
-+ if (info.inFormat == RTAUDIO_SINT8) {
-+ signed char *in = (signed char *)inBuffer;
-+ for (unsigned int i=0; i<stream_.bufferSize; i++) {
-+ for (j=0; j<info.channels; j++) {
-+ out[info.outOffset[j]] = (Int32) in[info.inOffset[j]];
-+ out[info.outOffset[j]] <<= 24;
-+ }
-+ in += info.inJump;
-+ out += info.outJump;
-+ }
-+ }
-+ else if (info.inFormat == RTAUDIO_SINT16) {
-+ Int16 *in = (Int16 *)inBuffer;
-+ for (unsigned int i=0; i<stream_.bufferSize; i++) {
-+ for (j=0; j<info.channels; j++) {
-+ out[info.outOffset[j]] = (Int32) in[info.inOffset[j]];
-+ out[info.outOffset[j]] <<= 16;
-+ }
-+ in += info.inJump;
-+ out += info.outJump;
-+ }
-+ }
-+ else if (info.inFormat == RTAUDIO_SINT24) {
-+ Int24 *in = (Int24 *)inBuffer;
-+ for (unsigned int i=0; i<stream_.bufferSize; i++) {
-+ for (j=0; j<info.channels; j++) {
-+ out[info.outOffset[j]] = (Int32) in[info.inOffset[j]].asInt();
-+ out[info.outOffset[j]] <<= 8;
-+ }
-+ in += info.inJump;
-+ out += info.outJump;
-+ }
-+ }
-+ else if (info.inFormat == RTAUDIO_SINT32) {
-+ // Channel compensation and/or (de)interleaving only.
-+ Int32 *in = (Int32 *)inBuffer;
-+ for (unsigned int i=0; i<stream_.bufferSize; i++) {
-+ for (j=0; j<info.channels; j++) {
-+ out[info.outOffset[j]] = in[info.inOffset[j]];
-+ }
-+ in += info.inJump;
-+ out += info.outJump;
-+ }
-+ }
-+ else if (info.inFormat == RTAUDIO_FLOAT32) {
-+ Float32 *in = (Float32 *)inBuffer;
-+ for (unsigned int i=0; i<stream_.bufferSize; i++) {
-+ for (j=0; j<info.channels; j++) {
-+ out[info.outOffset[j]] = (Int32) (in[info.inOffset[j]] * 2147483647.5 - 0.5);
-+ }
-+ in += info.inJump;
-+ out += info.outJump;
-+ }
-+ }
-+ else if (info.inFormat == RTAUDIO_FLOAT64) {
-+ Float64 *in = (Float64 *)inBuffer;
-+ for (unsigned int i=0; i<stream_.bufferSize; i++) {
-+ for (j=0; j<info.channels; j++) {
-+ out[info.outOffset[j]] = (Int32) (in[info.inOffset[j]] * 2147483647.5 - 0.5);
-+ }
-+ in += info.inJump;
-+ out += info.outJump;
-+ }
-+ }
-+ }
-+ else if (info.outFormat == RTAUDIO_SINT24) {
-+ Int24 *out = (Int24 *)outBuffer;
-+ if (info.inFormat == RTAUDIO_SINT8) {
-+ signed char *in = (signed char *)inBuffer;
-+ for (unsigned int i=0; i<stream_.bufferSize; i++) {
-+ for (j=0; j<info.channels; j++) {
-+ out[info.outOffset[j]] = (Int32) (in[info.inOffset[j]] << 16);
-+ //out[info.outOffset[j]] <<= 16;
-+ }
-+ in += info.inJump;
-+ out += info.outJump;
-+ }
-+ }
-+ else if (info.inFormat == RTAUDIO_SINT16) {
-+ Int16 *in = (Int16 *)inBuffer;
-+ for (unsigned int i=0; i<stream_.bufferSize; i++) {
-+ for (j=0; j<info.channels; j++) {
-+ out[info.outOffset[j]] = (Int32) (in[info.inOffset[j]] << 8);
-+ //out[info.outOffset[j]] <<= 8;
-+ }
-+ in += info.inJump;
-+ out += info.outJump;
-+ }
-+ }
-+ else if (info.inFormat == RTAUDIO_SINT24) {
-+ // Channel compensation and/or (de)interleaving only.
-+ Int24 *in = (Int24 *)inBuffer;
-+ for (unsigned int i=0; i<stream_.bufferSize; i++) {
-+ for (j=0; j<info.channels; j++) {
-+ out[info.outOffset[j]] = in[info.inOffset[j]];
-+ }
-+ in += info.inJump;
-+ out += info.outJump;
-+ }
-+ }
-+ else if (info.inFormat == RTAUDIO_SINT32) {
-+ Int32 *in = (Int32 *)inBuffer;
-+ for (unsigned int i=0; i<stream_.bufferSize; i++) {
-+ for (j=0; j<info.channels; j++) {
-+ out[info.outOffset[j]] = (Int32) (in[info.inOffset[j]] >> 8);
-+ //out[info.outOffset[j]] >>= 8;
-+ }
-+ in += info.inJump;
-+ out += info.outJump;
-+ }
-+ }
-+ else if (info.inFormat == RTAUDIO_FLOAT32) {
-+ Float32 *in = (Float32 *)inBuffer;
-+ for (unsigned int i=0; i<stream_.bufferSize; i++) {
-+ for (j=0; j<info.channels; j++) {
-+ out[info.outOffset[j]] = (Int32) (in[info.inOffset[j]] * 8388607.5 - 0.5);
-+ }
-+ in += info.inJump;
-+ out += info.outJump;
-+ }
-+ }
-+ else if (info.inFormat == RTAUDIO_FLOAT64) {
-+ Float64 *in = (Float64 *)inBuffer;
-+ for (unsigned int i=0; i<stream_.bufferSize; i++) {
-+ for (j=0; j<info.channels; j++) {
-+ out[info.outOffset[j]] = (Int32) (in[info.inOffset[j]] * 8388607.5 - 0.5);
-+ }
-+ in += info.inJump;
-+ out += info.outJump;
-+ }
-+ }
-+ }
-+ else if (info.outFormat == RTAUDIO_SINT16) {
-+ Int16 *out = (Int16 *)outBuffer;
-+ if (info.inFormat == RTAUDIO_SINT8) {
-+ signed char *in = (signed char *)inBuffer;
-+ for (unsigned int i=0; i<stream_.bufferSize; i++) {
-+ for (j=0; j<info.channels; j++) {
-+ out[info.outOffset[j]] = (Int16) in[info.inOffset[j]];
-+ out[info.outOffset[j]] <<= 8;
-+ }
-+ in += info.inJump;
-+ out += info.outJump;
-+ }
-+ }
-+ else if (info.inFormat == RTAUDIO_SINT16) {
-+ // Channel compensation and/or (de)interleaving only.
-+ Int16 *in = (Int16 *)inBuffer;
-+ for (unsigned int i=0; i<stream_.bufferSize; i++) {
-+ for (j=0; j<info.channels; j++) {
-+ out[info.outOffset[j]] = in[info.inOffset[j]];
-+ }
-+ in += info.inJump;
-+ out += info.outJump;
-+ }
-+ }
-+ else if (info.inFormat == RTAUDIO_SINT24) {
-+ Int24 *in = (Int24 *)inBuffer;
-+ for (unsigned int i=0; i<stream_.bufferSize; i++) {
-+ for (j=0; j<info.channels; j++) {
-+ out[info.outOffset[j]] = (Int16) (in[info.inOffset[j]].asInt() >> 8);
-+ }
-+ in += info.inJump;
-+ out += info.outJump;
-+ }
-+ }
-+ else if (info.inFormat == RTAUDIO_SINT32) {
-+ Int32 *in = (Int32 *)inBuffer;
-+ for (unsigned int i=0; i<stream_.bufferSize; i++) {
-+ for (j=0; j<info.channels; j++) {
-+ out[info.outOffset[j]] = (Int16) ((in[info.inOffset[j]] >> 16) & 0x0000ffff);
-+ }
-+ in += info.inJump;
-+ out += info.outJump;
-+ }
-+ }
-+ else if (info.inFormat == RTAUDIO_FLOAT32) {
-+ Float32 *in = (Float32 *)inBuffer;
-+ for (unsigned int i=0; i<stream_.bufferSize; i++) {
-+ for (j=0; j<info.channels; j++) {
-+ out[info.outOffset[j]] = (Int16) (in[info.inOffset[j]] * 32767.5 - 0.5);
-+ }
-+ in += info.inJump;
-+ out += info.outJump;
-+ }
-+ }
-+ else if (info.inFormat == RTAUDIO_FLOAT64) {
-+ Float64 *in = (Float64 *)inBuffer;
-+ for (unsigned int i=0; i<stream_.bufferSize; i++) {
-+ for (j=0; j<info.channels; j++) {
-+ out[info.outOffset[j]] = (Int16) (in[info.inOffset[j]] * 32767.5 - 0.5);
-+ }
-+ in += info.inJump;
-+ out += info.outJump;
-+ }
-+ }
-+ }
-+ else if (info.outFormat == RTAUDIO_SINT8) {
-+ signed char *out = (signed char *)outBuffer;
-+ if (info.inFormat == RTAUDIO_SINT8) {
-+ // Channel compensation and/or (de)interleaving only.
-+ signed char *in = (signed char *)inBuffer;
-+ for (unsigned int i=0; i<stream_.bufferSize; i++) {
-+ for (j=0; j<info.channels; j++) {
-+ out[info.outOffset[j]] = in[info.inOffset[j]];
-+ }
-+ in += info.inJump;
-+ out += info.outJump;
-+ }
-+ }
-+ if (info.inFormat == RTAUDIO_SINT16) {
-+ Int16 *in = (Int16 *)inBuffer;
-+ for (unsigned int i=0; i<stream_.bufferSize; i++) {
-+ for (j=0; j<info.channels; j++) {
-+ out[info.outOffset[j]] = (signed char) ((in[info.inOffset[j]] >> 8) & 0x00ff);
-+ }
-+ in += info.inJump;
-+ out += info.outJump;
-+ }
-+ }
-+ else if (info.inFormat == RTAUDIO_SINT24) {
-+ Int24 *in = (Int24 *)inBuffer;
-+ for (unsigned int i=0; i<stream_.bufferSize; i++) {
-+ for (j=0; j<info.channels; j++) {
-+ out[info.outOffset[j]] = (signed char) (in[info.inOffset[j]].asInt() >> 16);
-+ }
-+ in += info.inJump;
-+ out += info.outJump;
-+ }
-+ }
-+ else if (info.inFormat == RTAUDIO_SINT32) {
-+ Int32 *in = (Int32 *)inBuffer;
-+ for (unsigned int i=0; i<stream_.bufferSize; i++) {
-+ for (j=0; j<info.channels; j++) {
-+ out[info.outOffset[j]] = (signed char) ((in[info.inOffset[j]] >> 24) & 0x000000ff);
-+ }
-+ in += info.inJump;
-+ out += info.outJump;
-+ }
-+ }
-+ else if (info.inFormat == RTAUDIO_FLOAT32) {
-+ Float32 *in = (Float32 *)inBuffer;
-+ for (unsigned int i=0; i<stream_.bufferSize; i++) {
-+ for (j=0; j<info.channels; j++) {
-+ out[info.outOffset[j]] = (signed char) (in[info.inOffset[j]] * 127.5 - 0.5);
-+ }
-+ in += info.inJump;
-+ out += info.outJump;
-+ }
-+ }
-+ else if (info.inFormat == RTAUDIO_FLOAT64) {
-+ Float64 *in = (Float64 *)inBuffer;
-+ for (unsigned int i=0; i<stream_.bufferSize; i++) {
-+ for (j=0; j<info.channels; j++) {
-+ out[info.outOffset[j]] = (signed char) (in[info.inOffset[j]] * 127.5 - 0.5);
-+ }
-+ in += info.inJump;
-+ out += info.outJump;
-+ }
-+ }
-+ }
-+}
-+
-+//static inline uint16_t bswap_16(uint16_t x) { return (x>>8) | (x<<8); }
-+//static inline uint32_t bswap_32(uint32_t x) { return (bswap_16(x&0xffff)<<16) | (bswap_16(x>>16)); }
-+//static inline uint64_t bswap_64(uint64_t x) { return (((unsigned long long)bswap_32(x&0xffffffffull))<<32) | (bswap_32(x>>32)); }
-+
-+void RtApi :: byteSwapBuffer( char *buffer, unsigned int samples, RtAudioFormat format )
-+{
-+ char val;
-+ char *ptr;
-+
-+ ptr = buffer;
-+ if ( format == RTAUDIO_SINT16 ) {
-+ for ( unsigned int i=0; i<samples; i++ ) {
-+ // Swap 1st and 2nd bytes.
-+ val = *(ptr);
-+ *(ptr) = *(ptr+1);
-+ *(ptr+1) = val;
-+
-+ // Increment 2 bytes.
-+ ptr += 2;
-+ }
-+ }
-+ else if ( format == RTAUDIO_SINT32 ||
-+ format == RTAUDIO_FLOAT32 ) {
-+ for ( unsigned int i=0; i<samples; i++ ) {
-+ // Swap 1st and 4th bytes.
-+ val = *(ptr);
-+ *(ptr) = *(ptr+3);
-+ *(ptr+3) = val;
-+
-+ // Swap 2nd and 3rd bytes.
-+ ptr += 1;
-+ val = *(ptr);
-+ *(ptr) = *(ptr+1);
-+ *(ptr+1) = val;
-+
-+ // Increment 3 more bytes.
-+ ptr += 3;
-+ }
-+ }
-+ else if ( format == RTAUDIO_SINT24 ) {
-+ for ( unsigned int i=0; i<samples; i++ ) {
-+ // Swap 1st and 3rd bytes.
-+ val = *(ptr);
-+ *(ptr) = *(ptr+2);
-+ *(ptr+2) = val;
-+
-+ // Increment 2 more bytes.
-+ ptr += 2;
-+ }
-+ }
-+ else if ( format == RTAUDIO_FLOAT64 ) {
-+ for ( unsigned int i=0; i<samples; i++ ) {
-+ // Swap 1st and 8th bytes
-+ val = *(ptr);
-+ *(ptr) = *(ptr+7);
-+ *(ptr+7) = val;
-+
-+ // Swap 2nd and 7th bytes
-+ ptr += 1;
-+ val = *(ptr);
-+ *(ptr) = *(ptr+5);
-+ *(ptr+5) = val;
-+
-+ // Swap 3rd and 6th bytes
-+ ptr += 1;
-+ val = *(ptr);
-+ *(ptr) = *(ptr+3);
-+ *(ptr+3) = val;
-+
-+ // Swap 4th and 5th bytes
-+ ptr += 1;
-+ val = *(ptr);
-+ *(ptr) = *(ptr+1);
-+ *(ptr+1) = val;
-+
-+ // Increment 5 more bytes.
-+ ptr += 5;
-+ }
-+ }
-+}
-+
-+void *RtAudio :: GIADA_HACK__getJackClient() { /* Monocasual HACK */
-+#if defined(__UNIX_JACK__)
-+ RtApiJack*jackapi = dynamic_cast<RtApiJack*>(rtapi_);
-+ if (jackapi && jackapi->stream_.apiHandle) {
-+ JackHandle *handle = (JackHandle *) jackapi->stream_.apiHandle;
-+ return (void*) handle->client;
-+ }
-+#endif
-+ return 0;
-+}
-+
-+
-+
-+ // Indentation settings for Vim and Emacs
-+ //
-+ // Local Variables:
-+ // c-basic-offset: 2
-+ // indent-tabs-mode: nil
-+ // End:
-+ //
-+ // vim: et sts=2 sw=2
-+
---- giada.orig/src/deps/rtaudio-mod/RtAudio.h
-+++ giada/src/deps/rtaudio-mod/RtAudio.h
-@@ -10,7 +10,7 @@
- RtAudio WWW site: http://www.music.mcgill.ca/~gary/rtaudio/
-
- RtAudio: realtime audio i/o C++ classes
-- Copyright (c) 2001-2016 Gary P. Scavone
-+ 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
-@@ -45,11 +45,11 @@
- #ifndef __RTAUDIO_H
- #define __RTAUDIO_H
-
--#define RTAUDIO_VERSION "4.1.2"
-+#define RTAUDIO_VERSION "5.0.0"
-
- #include <string>
- #include <vector>
--#include <exception>
-+#include <stdexcept>
- #include <iostream>
-
- /*! \typedef typedef unsigned long RtAudioFormat;
-@@ -86,6 +86,7 @@
- - \e RTAUDIO_MINIMIZE_LATENCY: Attempt to set stream parameters for lowest possible latency.
- - \e RTAUDIO_HOG_DEVICE: Attempt grab device for exclusive use.
- - \e RTAUDIO_ALSA_USE_DEFAULT: Use the "default" PCM device (ALSA only).
-+ - \e RTAUDIO_JACK_DONT_CONNECT: Do not automatically connect ports (JACK only).
-
- By default, RtAudio streams pass and receive audio data from the
- client in an interleaved format. By passing the
-@@ -111,12 +112,15 @@
- open the input and/or output stream device(s) for exclusive use.
- Note that this is not possible with all supported audio APIs.
-
-- If the RTAUDIO_SCHEDULE_REALTIME flag is set, RtAudio will attempt
-+ If the RTAUDIO_SCHEDULE_REALTIME flag is set, RtAudio will attempt
- to select realtime scheduling (round-robin) for the callback thread.
-
- If the RTAUDIO_ALSA_USE_DEFAULT flag is set, RtAudio will attempt to
- open the "default" PCM device when using the ALSA API. Note that this
- will override any specified input or output device id.
-+
-+ If the RTAUDIO_JACK_DONT_CONNECT flag is set, RtAudio will not attempt
-+ to automatically connect the ports of the client to the audio device.
- */
- typedef unsigned int RtAudioStreamFlags;
- static const RtAudioStreamFlags RTAUDIO_NONINTERLEAVED = 0x1; // Use non-interleaved buffers (default = interleaved).
-@@ -124,6 +128,7 @@
- static const RtAudioStreamFlags RTAUDIO_HOG_DEVICE = 0x4; // Attempt grab device and prevent use by others.
- static const RtAudioStreamFlags RTAUDIO_SCHEDULE_REALTIME = 0x8; // Try to select realtime scheduling for callback thread.
- static const RtAudioStreamFlags RTAUDIO_ALSA_USE_DEFAULT = 0x10; // Use the "default" PCM device (ALSA only).
-+static const RtAudioStreamFlags RTAUDIO_JACK_DONT_CONNECT = 0x20; // Do not automatically connect ports (JACK only).
-
- /*! \typedef typedef unsigned long RtAudioStreamStatus;
- \brief RtAudio stream status (over- or underflow) flags.
-@@ -195,7 +200,7 @@
- */
- /************************************************************************/
-
--class RtAudioError : public std::exception
-+class RtAudioError : public std::runtime_error
- {
- public:
- //! Defined RtAudioError types.
-@@ -214,25 +219,22 @@
- };
-
- //! The constructor.
-- RtAudioError( const std::string& message, Type type = RtAudioError::UNSPECIFIED ) throw() : message_(message), type_(type) {}
--
-- //! The destructor.
-- virtual ~RtAudioError( void ) throw() {}
-+ RtAudioError( const std::string& message,
-+ Type type = RtAudioError::UNSPECIFIED )
-+ : std::runtime_error(message), type_(type) {}
-
- //! Prints thrown error message to stderr.
-- virtual void printMessage( void ) const throw() { std::cerr << '\n' << message_ << "\n\n"; }
-+ virtual void printMessage( void ) const
-+ { std::cerr << '\n' << what() << "\n\n"; }
-
- //! Returns the thrown error message type.
-- virtual const Type& getType(void) const throw() { return type_; }
-+ virtual const Type& getType(void) const { return type_; }
-
- //! Returns the thrown error message string.
-- virtual const std::string& getMessage(void) const throw() { return message_; }
--
-- //! Returns the thrown error message as a c-style string.
-- virtual const char* what( void ) const throw() { return message_.c_str(); }
-+ virtual const std::string getMessage(void) const
-+ { return std::string(what()); }
-
- protected:
-- std::string message_;
- Type type_;
- };
-
-@@ -341,7 +343,7 @@
- open the input and/or output stream device(s) for exclusive use.
- Note that this is not possible with all supported audio APIs.
-
-- If the RTAUDIO_SCHEDULE_REALTIME flag is set, RtAudio will attempt
-+ If the RTAUDIO_SCHEDULE_REALTIME flag is set, RtAudio will attempt
- to select realtime scheduling (round-robin) for the callback thread.
- The \c priority parameter will only be used if the RTAUDIO_SCHEDULE_REALTIME
- flag is set. It defines the thread's realtime priority.
-@@ -375,7 +377,7 @@
- };
-
- //! A static function to determine the current RtAudio version.
-- static std::string getVersion( void ) throw();
-+ static std::string getVersion( void );
-
- //! A static function to determine the available compiled audio APIs.
- /*!
-@@ -383,7 +385,7 @@
- the enumerated list values. Note that there can be more than one
- API compiled for certain operating systems.
- */
-- static void getCompiledApi( std::vector<RtAudio::Api> &apis ) throw();
-+ static void getCompiledApi( std::vector<RtAudio::Api> &apis );
-
- //! The class constructor.
- /*!
-@@ -401,18 +403,18 @@
- If a stream is running or open, it will be stopped and closed
- automatically.
- */
-- ~RtAudio() throw();
-+ ~RtAudio();
-
- //! Returns the audio API specifier for the current instance of RtAudio.
-- RtAudio::Api getCurrentApi( void ) throw();
-+ RtAudio::Api getCurrentApi( void );
-
- //! A public function that queries for the number of audio devices available.
- /*!
- This function performs a system query of available devices each time it
- is called, thus supporting devices connected \e after instantiation. If
-- a system error occurs during processing, a warning will be issued.
-+ a system error occurs during processing, a warning will be issued.
- */
-- unsigned int getDeviceCount( void ) throw();
-+ unsigned int getDeviceCount( void );
-
- //! Return an RtAudio::DeviceInfo structure for a specified device number.
- /*!
-@@ -435,7 +437,7 @@
- client's responsibility to verify that a device is available
- before attempting to open a stream.
- */
-- unsigned int getDefaultOutputDevice( void ) throw();
-+ unsigned int getDefaultOutputDevice( void );
-
- //! A function that returns the index of the default input device.
- /*!
-@@ -445,7 +447,7 @@
- client's responsibility to verify that a device is available
- before attempting to open a stream.
- */
-- unsigned int getDefaultInputDevice( void ) throw();
-+ unsigned int getDefaultInputDevice( void );
-
- //! A public function for opening a stream with the specified parameters.
- /*!
-@@ -477,7 +479,7 @@
- from within the callback function.
- \param options An optional pointer to a structure containing various
- global stream options, including a list of OR'ed RtAudioStreamFlags
-- and a suggested number of stream buffers that can be used to
-+ and a suggested number of stream buffers that can be used to
- control stream latency. More buffers typically result in more
- robust performance, though at a cost of greater latency. If a
- value of zero is specified, a system-specific median value is
-@@ -498,7 +500,7 @@
- If a stream is not open, this function issues a warning and
- returns (no exception is thrown).
- */
-- void closeStream( void ) throw();
-+ void closeStream( void );
-
- //! A function that starts a stream.
- /*!
-@@ -528,10 +530,10 @@
- void abortStream( void );
-
- //! Returns true if a stream is open and false if not.
-- bool isStreamOpen( void ) const throw();
-+ bool isStreamOpen( void ) const;
-
- //! Returns true if the stream is running and false if it is stopped or not open.
-- bool isStreamRunning( void ) const throw();
-+ bool isStreamRunning( void ) const;
-
- //! Returns the number of elapsed seconds since the stream was started.
- /*!
-@@ -565,14 +567,15 @@
- unsigned int getStreamSampleRate( void );
-
- //! Specify whether warning messages should be printed to stderr.
-- void showWarnings( bool value = true ) throw();
-+ void showWarnings( bool value = true );
-
-- /* --- Monocasual hack ---------------------------------------------------- */
-- //protected:
-- /* ------------------------------------------------------------------------ */
-+ protected:
-
- void openRtApi( RtAudio::Api api );
- RtApi *rtapi_;
-+
-+ public:
-+ void *GIADA_HACK__getJackClient(); /* Monocasual HACK */
- };
-
- // Operating system dependent thread functionality.
-@@ -618,7 +621,7 @@
-
- // Default constructor.
- CallbackInfo()
-- :object(0), callback(0), userData(0), errorCallback(0), apiInfo(0), isRunning(false), doRealtime(false) {}
-+ :object(0), callback(0), userData(0), errorCallback(0), apiInfo(0), isRunning(false), doRealtime(false), priority(0) {}
- };
-
- // **************************************************************** //
-@@ -675,12 +678,6 @@
- {
- public:
-
-- /* --- Monocasual hack ---------------------------------------------------- */
-- #if defined(__linux__) || defined(__FreeBSD__)
-- void *__HACK__getJackClient();
-- #endif
-- /* ------------------------------------------------------------------------ */
--
- RtApi();
- virtual ~RtApi();
- virtual RtAudio::Api getCurrentApi( void ) = 0;
-@@ -790,7 +787,7 @@
- "warning" message is reported and FAILURE is returned. A
- successful probe is indicated by a return value of SUCCESS.
- */
-- virtual bool probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,
-+ virtual bool probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,
- unsigned int firstChannel, unsigned int sampleRate,
- RtAudioFormat format, unsigned int *bufferSize,
- RtAudio::StreamOptions *options );
-@@ -824,6 +821,8 @@
-
- //! Protected common method that sets up the parameters for buffer conversion.
- void setConvertInfo( StreamMode mode, unsigned int firstChannel );
-+
-+ friend class RtAudio; /* GIADA Hack */
- };
-
- // **************************************************************** //
-@@ -832,22 +831,22 @@
- //
- // **************************************************************** //
-
--inline RtAudio::Api RtAudio :: getCurrentApi( void ) throw() { return rtapi_->getCurrentApi(); }
--inline unsigned int RtAudio :: getDeviceCount( void ) throw() { return rtapi_->getDeviceCount(); }
-+inline RtAudio::Api RtAudio :: getCurrentApi( void ) { return rtapi_->getCurrentApi(); }
-+inline unsigned int RtAudio :: getDeviceCount( void ) { return rtapi_->getDeviceCount(); }
- inline RtAudio::DeviceInfo RtAudio :: getDeviceInfo( unsigned int device ) { return rtapi_->getDeviceInfo( device ); }
--inline unsigned int RtAudio :: getDefaultInputDevice( void ) throw() { return rtapi_->getDefaultInputDevice(); }
--inline unsigned int RtAudio :: getDefaultOutputDevice( void ) throw() { return rtapi_->getDefaultOutputDevice(); }
--inline void RtAudio :: closeStream( void ) throw() { return rtapi_->closeStream(); }
-+inline unsigned int RtAudio :: getDefaultInputDevice( void ) { return rtapi_->getDefaultInputDevice(); }
-+inline unsigned int RtAudio :: getDefaultOutputDevice( void ) { return rtapi_->getDefaultOutputDevice(); }
-+inline void RtAudio :: closeStream( void ) { return rtapi_->closeStream(); }
- inline void RtAudio :: startStream( void ) { return rtapi_->startStream(); }
- inline void RtAudio :: stopStream( void ) { return rtapi_->stopStream(); }
- inline void RtAudio :: abortStream( void ) { return rtapi_->abortStream(); }
--inline bool RtAudio :: isStreamOpen( void ) const throw() { return rtapi_->isStreamOpen(); }
--inline bool RtAudio :: isStreamRunning( void ) const throw() { return rtapi_->isStreamRunning(); }
-+inline bool RtAudio :: isStreamOpen( void ) const { return rtapi_->isStreamOpen(); }
-+inline bool RtAudio :: isStreamRunning( void ) const { return rtapi_->isStreamRunning(); }
- inline long RtAudio :: getStreamLatency( void ) { return rtapi_->getStreamLatency(); }
- inline unsigned int RtAudio :: getStreamSampleRate( void ) { return rtapi_->getStreamSampleRate(); }
- inline double RtAudio :: getStreamTime( void ) { return rtapi_->getStreamTime(); }
- inline void RtAudio :: setStreamTime( double time ) { return rtapi_->setStreamTime( time ); }
--inline void RtAudio :: showWarnings( bool value ) throw() { rtapi_->showWarnings( value ); }
-+inline void RtAudio :: showWarnings( bool value ) { rtapi_->showWarnings( value ); }
-
- // RtApi Subclass prototypes.
-
-@@ -882,7 +881,7 @@
-
- private:
-
-- bool probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,
-+ bool probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,
- unsigned int firstChannel, unsigned int sampleRate,
- RtAudioFormat format, unsigned int *bufferSize,
- RtAudio::StreamOptions *options );
-@@ -916,10 +915,12 @@
-
- private:
-
-- bool probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,
-+ bool probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,
- unsigned int firstChannel, unsigned int sampleRate,
- RtAudioFormat format, unsigned int *bufferSize,
- RtAudio::StreamOptions *options );
-+
-+ bool shouldAutoconnect_;
- };
-
- #endif
-@@ -952,7 +953,7 @@
- std::vector<RtAudio::DeviceInfo> devices_;
- void saveDeviceInfo( void );
- bool coInitialized_;
-- bool probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,
-+ bool probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,
- unsigned int firstChannel, unsigned int sampleRate,
- RtAudioFormat format, unsigned int *bufferSize,
- RtAudio::StreamOptions *options );
-@@ -991,7 +992,7 @@
- bool buffersRolling;
- long duplexPrerollBytes;
- std::vector<struct DsDevice> dsDevices;
-- bool probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,
-+ bool probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,
- unsigned int firstChannel, unsigned int sampleRate,
- RtAudioFormat format, unsigned int *bufferSize,
- RtAudio::StreamOptions *options );
-@@ -1062,7 +1063,7 @@
-
- std::vector<RtAudio::DeviceInfo> devices_;
- void saveDeviceInfo( void );
-- bool probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,
-+ bool probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,
- unsigned int firstChannel, unsigned int sampleRate,
- RtAudioFormat format, unsigned int *bufferSize,
- RtAudio::StreamOptions *options );
-@@ -1126,7 +1127,7 @@
-
- private:
-
-- bool probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,
-+ bool probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,
- unsigned int firstChannel, unsigned int sampleRate,
- RtAudioFormat format, unsigned int *bufferSize,
- RtAudio::StreamOptions *options );
-@@ -1151,7 +1152,7 @@
-
- private:
-
-- bool probeDeviceOpen( unsigned int /*device*/, StreamMode /*mode*/, unsigned int /*channels*/,
-+ bool probeDeviceOpen( unsigned int /*device*/, StreamMode /*mode*/, unsigned int /*channels*/,
- unsigned int /*firstChannel*/, unsigned int /*sampleRate*/,
- RtAudioFormat /*format*/, unsigned int * /*bufferSize*/,
- RtAudio::StreamOptions * /*options*/ ) { return false; }
projects / giada.git / commitdiff