--- /dev/null
--- /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
++@@ -59,7 +59,7 @@
++
++ 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
++-#ifdef __linux__\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 ---------------------------------------------------- */
++- #ifdef __linux__
++- 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
