pool/libmlt
SHA256
6
0
Fork 1
Code Issues Pull Requests Packages Projects Releases Wiki Activity
libmlt/0001-fix-930-support-RtAudio-6.patch

23417 lines
852 KiB
Diff
Raw Blame History

From 80946f7aa6c31b5c6017c2d33e91fb5447562bae Mon Sep 17 00:00:00 2001
From: Dan Dennedy <dan@dennedy.org>
Date: Fri, 13 Oct 2023 00:04:04 -0700
Subject: [PATCH] fix #930 support RtAudio 6
---
src/modules/rtaudio/LICENSE | 27 +
src/modules/rtaudio/RtAudio.cpp | 21757 +++++++++++----------
src/modules/rtaudio/RtAudio.h | 874 +-
src/modules/rtaudio/consumer_rtaudio.cpp | 136 +-
4 files changed, 11974 insertions(+), 10820 deletions(-)
create mode 100644 src/modules/rtaudio/LICENSE
diff --git a/src/modules/rtaudio/LICENSE b/src/modules/rtaudio/LICENSE
new file mode 100644
index 00000000..fcb119e5
--- /dev/null
+++ b/src/modules/rtaudio/LICENSE
@@ -0,0 +1,27 @@
+
+RtAudio: a set of realtime audio i/o C++ classes
+Copyright (c) 2001-2023 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.
diff --git a/src/modules/rtaudio/RtAudio.cpp b/src/modules/rtaudio/RtAudio.cpp
index 50ee333f..b13f04eb 100644
--- a/src/modules/rtaudio/RtAudio.cpp
+++ b/src/modules/rtaudio/RtAudio.cpp
@@ -1,10230 +1,11527 @@
-/************************************************************************/
-/*! \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) 2001-2016 Gary P. Scavone
-
- Permission is hereby granted, free of charge, to any person
- obtaining a copy of this software and associated documentation files
- (the "Software"), to deal in the Software without restriction,
- including without limitation the rights to use, copy, modify, merge,
- publish, distribute, sublicense, and/or sell copies of the Software,
- and to permit persons to whom the Software is furnished to do so,
- subject to the following conditions:
-
- The above copyright notice and this permission notice shall be
- included in all copies or substantial portions of the Software.
-
- Any person wishing to distribute modifications to the Software is
- asked to send the modifications to the original developer so that
- they can be incorporated into the canonical version. This is,
- however, not a binding provision of this license.
-
- THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND,
- EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF
- MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT.
- IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR
- ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF
- CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION
- WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
-*/
-/************************************************************************/
-
-// RtAudio: Version 4.1.2
-
-#include "RtAudio.h"
-#include <iostream>
-#include <cstdlib>
-#include <cstring>
-#include <climits>
-#include <algorithm>
-
-// Static variable definitions.
-const unsigned int RtApi::MAX_SAMPLE_RATES = 14;
-const unsigned int RtApi::SAMPLE_RATES[] = {
- 4000, 5512, 8000, 9600, 11025, 16000, 22050,
- 32000, 44100, 48000, 88200, 96000, 176400, 192000
-};
-
-#if defined(__WINDOWS_DS__) || defined(__WINDOWS_ASIO__) || defined(__WINDOWS_WASAPI__)
- #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 ) throw()
-{
- return RTAUDIO_VERSION;
-}
-
-void RtAudio :: getCompiledApi( std::vector<RtAudio::Api> &apis ) throw()
-{
- apis.clear();
-
- // The order here will control the order of RtAudio's API search in
- // the constructor.
-#if defined(__UNIX_JACK__)
- apis.push_back( UNIX_JACK );
-#endif
-#if defined(__LINUX_ALSA__)
- apis.push_back( LINUX_ALSA );
-#endif
-#if defined(__LINUX_PULSE__)
- apis.push_back( LINUX_PULSE );
-#endif
-#if defined(__LINUX_OSS__)
- apis.push_back( LINUX_OSS );
-#endif
-#if defined(__WINDOWS_ASIO__)
- apis.push_back( WINDOWS_ASIO );
-#endif
-#if defined(__WINDOWS_WASAPI__)
- apis.push_back( WINDOWS_WASAPI );
-#endif
-#if defined(__WINDOWS_DS__)
- apis.push_back( WINDOWS_DS );
-#endif
-#if defined(__MACOSX_CORE__)
- apis.push_back( MACOSX_CORE );
-#endif
-#if defined(__RTAUDIO_DUMMY__)
- apis.push_back( RTAUDIO_DUMMY );
-#endif
-}
-
-void RtAudio :: openRtApi( RtAudio::Api api )
-{
- if ( rtapi_ )
- delete rtapi_;
- rtapi_ = 0;
-
-#if defined(__UNIX_JACK__)
- if ( api == UNIX_JACK )
- rtapi_ = new RtApiJack();
-#endif
-#if defined(__LINUX_ALSA__)
- if ( api == LINUX_ALSA )
- rtapi_ = new RtApiAlsa();
-#endif
-#if defined(__LINUX_PULSE__)
- if ( api == LINUX_PULSE )
- rtapi_ = new RtApiPulse();
-#endif
-#if defined(__LINUX_OSS__)
- if ( api == LINUX_OSS )
- rtapi_ = new RtApiOss();
-#endif
-#if defined(__WINDOWS_ASIO__)
- if ( api == WINDOWS_ASIO )
- rtapi_ = new RtApiAsio();
-#endif
-#if defined(__WINDOWS_WASAPI__)
- if ( api == WINDOWS_WASAPI )
- rtapi_ = new RtApiWasapi();
-#endif
-#if defined(__WINDOWS_DS__)
- if ( api == WINDOWS_DS )
- rtapi_ = new RtApiDs();
-#endif
-#if defined(__MACOSX_CORE__)
- if ( api == MACOSX_CORE )
- rtapi_ = new RtApiCore();
-#endif
-#if defined(__RTAUDIO_DUMMY__)
- if ( api == RTAUDIO_DUMMY )
- rtapi_ = new RtApiDummy();
-#endif
-}
-
-RtAudio :: RtAudio( RtAudio::Api api )
-{
- rtapi_ = 0;
-
- if ( api != UNSPECIFIED ) {
- // Attempt to open the specified API.
- openRtApi( api );
- if ( rtapi_ ) return;
-
- // No compiled support for specified API value. Issue a debug
- // warning and continue as if no API was specified.
- std::cerr << "\nRtAudio: no compiled support for specified API argument!\n" << std::endl;
- }
-
- // Iterate through the compiled APIs and return as soon as we find
- // one with at least one device or we reach the end of the list.
- std::vector< RtAudio::Api > apis;
- getCompiledApi( apis );
- for ( unsigned int i=0; i<apis.size(); i++ ) {
- openRtApi( apis[i] );
- if ( rtapi_ && rtapi_->getDeviceCount() ) break;
- }
-
- if ( rtapi_ ) return;
-
- // It should not be possible to get here because the preprocessor
- // definition __RTAUDIO_DUMMY__ is automatically defined if no
- // API-specific definitions are passed to the compiler. But just in
- // case something weird happens, we'll throw an error.
- std::string errorText = "\nRtAudio: no compiled API support found ... critical error!!\n\n";
- throw( RtAudioError( errorText, RtAudioError::UNSPECIFIED ) );
-}
-
-RtAudio :: ~RtAudio() throw()
-{
- if ( rtapi_ )
- delete rtapi_;
-}
-
-void RtAudio :: openStream( RtAudio::StreamParameters *outputParameters,
- RtAudio::StreamParameters *inputParameters,
- RtAudioFormat format, unsigned int sampleRate,
- unsigned int *bufferFrames,
- RtAudioCallback callback, void *userData,
- RtAudio::StreamOptions *options,
- RtAudioErrorCallback errorCallback )
-{
- return rtapi_->openStream( outputParameters, inputParameters, format,
- sampleRate, bufferFrames, callback,
- userData, options, errorCallback );
-}
-
-// *************************************************** //
-//
-// Public RtApi definitions (see end of file for
-// private or protected utility functions).
-//
-// *************************************************** //
-
-RtApi :: RtApi()
-{
- stream_.state = STREAM_CLOSED;
- stream_.mode = UNINITIALIZED;
- stream_.apiHandle = 0;
- stream_.userBuffer[0] = 0;
- stream_.userBuffer[1] = 0;
- MUTEX_INITIALIZE( &stream_.mutex );
- showWarnings_ = true;
- firstErrorOccurred_ = false;
-}
-
-RtApi :: ~RtApi()
-{
- MUTEX_DESTROY( &stream_.mutex );
-}
-
-void RtApi :: openStream( RtAudio::StreamParameters *oParams,
- RtAudio::StreamParameters *iParams,
- RtAudioFormat format, unsigned int sampleRate,
- unsigned int *bufferFrames,
- RtAudioCallback callback, void *userData,
- RtAudio::StreamOptions *options,
- RtAudioErrorCallback errorCallback )
-{
- if ( stream_.state != STREAM_CLOSED ) {
- errorText_ = "RtApi::openStream: a stream is already open!";
- error( RtAudioError::INVALID_USE );
- return;
- }
-
- // Clear stream information potentially left from a previously open stream.
- clearStreamInfo();
-
- if ( oParams && oParams->nChannels < 1 ) {
- errorText_ = "RtApi::openStream: a non-NULL output StreamParameters structure cannot have an nChannels value less than one.";
- error( RtAudioError::INVALID_USE );
- return;
- }
-
- if ( iParams && iParams->nChannels < 1 ) {
- errorText_ = "RtApi::openStream: a non-NULL input StreamParameters structure cannot have an nChannels value less than one.";
- error( RtAudioError::INVALID_USE );
- return;
- }
-
- if ( oParams == NULL && iParams == NULL ) {
- errorText_ = "RtApi::openStream: input and output StreamParameters structures are both NULL!";
- error( RtAudioError::INVALID_USE );
- return;
- }
-
- if ( formatBytes(format) == 0 ) {
- errorText_ = "RtApi::openStream: 'format' parameter value is undefined.";
- error( RtAudioError::INVALID_USE );
- return;
- }
-
- unsigned int nDevices = getDeviceCount();
- unsigned int oChannels = 0;
- if ( oParams ) {
- oChannels = oParams->nChannels;
- if ( oParams->deviceId >= nDevices ) {
- errorText_ = "RtApi::openStream: output device parameter value is invalid.";
- error( RtAudioError::INVALID_USE );
- return;
- }
- }
-
- unsigned int iChannels = 0;
- if ( iParams ) {
- iChannels = iParams->nChannels;
- if ( iParams->deviceId >= nDevices ) {
- errorText_ = "RtApi::openStream: input device parameter value is invalid.";
- error( RtAudioError::INVALID_USE );
- return;
- }
- }
-
- bool result;
-
- if ( oChannels > 0 ) {
-
- result = probeDeviceOpen( oParams->deviceId, OUTPUT, oChannels, oParams->firstChannel,
- sampleRate, format, bufferFrames, options );
- if ( result == false ) {
- error( RtAudioError::SYSTEM_ERROR );
- return;
- }
- }
-
- if ( iChannels > 0 ) {
-
- result = probeDeviceOpen( iParams->deviceId, INPUT, iChannels, iParams->firstChannel,
- sampleRate, format, bufferFrames, options );
- if ( result == false ) {
- if ( oChannels > 0 ) closeStream();
- error( RtAudioError::SYSTEM_ERROR );
- return;
- }
- }
-
- stream_.callbackInfo.callback = (void *) callback;
- stream_.callbackInfo.userData = userData;
- stream_.callbackInfo.errorCallback = (void *) errorCallback;
-
- if ( options ) options->numberOfBuffers = stream_.nBuffers;
- stream_.state = STREAM_STOPPED;
-}
-
-unsigned int RtApi :: getDefaultInputDevice( void )
-{
- // Should be implemented in subclasses if possible.
- return 0;
-}
-
-unsigned int RtApi :: getDefaultOutputDevice( void )
-{
- // Should be implemented in subclasses if possible.
- return 0;
-}
-
-void RtApi :: closeStream( void )
-{
- // MUST be implemented in subclasses!
- return;
-}
-
-bool RtApi :: probeDeviceOpen( unsigned int /*device*/, StreamMode /*mode*/, unsigned int /*channels*/,
- unsigned int /*firstChannel*/, unsigned int /*sampleRate*/,
- RtAudioFormat /*format*/, unsigned int * /*bufferSize*/,
- RtAudio::StreamOptions * /*options*/ )
-{
- // MUST be implemented in subclasses!
- return FAILURE;
-}
-
-void RtApi :: tickStreamTime( void )
-{
- // Subclasses that do not provide their own implementation of
- // getStreamTime should call this function once per buffer I/O to
- // provide basic stream time support.
-
- stream_.streamTime += ( stream_.bufferSize * 1.0 / stream_.sampleRate );
-
-#if defined( HAVE_GETTIMEOFDAY )
- gettimeofday( &stream_.lastTickTimestamp, NULL );
-#endif
-}
-
-long RtApi :: getStreamLatency( void )
-{
- verifyStream();
-
- long totalLatency = 0;
- if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX )
- totalLatency = stream_.latency[0];
- if ( stream_.mode == INPUT || stream_.mode == DUPLEX )
- totalLatency += stream_.latency[1];
-
- return totalLatency;
-}
-
-double RtApi :: getStreamTime( void )
-{
- verifyStream();
-
-#if defined( HAVE_GETTIMEOFDAY )
- // Return a very accurate estimate of the stream time by
- // adding in the elapsed time since the last tick.
- struct timeval then;
- struct timeval now;
-
- if ( stream_.state != STREAM_RUNNING || stream_.streamTime == 0.0 )
- return stream_.streamTime;
-
- gettimeofday( &now, NULL );
- then = stream_.lastTickTimestamp;
- return stream_.streamTime +
- ((now.tv_sec + 0.000001 * now.tv_usec) -
- (then.tv_sec + 0.000001 * then.tv_usec));
-#else
- return stream_.streamTime;
-#endif
-}
-
-void RtApi :: setStreamTime( double time )
-{
- verifyStream();
-
- if ( time >= 0.0 )
- stream_.streamTime = time;
-}
-
-unsigned int RtApi :: getStreamSampleRate( void )
-{
- verifyStream();
-
- return stream_.sampleRate;
-}
-
-
-// *************************************************** //
-//
-// OS/API-specific methods.
-//
-// *************************************************** //
-
-#if defined(__MACOSX_CORE__)
-
-// The OS X CoreAudio API is designed to use a separate callback
-// procedure for each of its audio devices. A single RtAudio duplex
-// stream using two different devices is supported here, though it
-// cannot be guaranteed to always behave correctly because we cannot
-// synchronize these two callbacks.
-//
-// A property listener is installed for over/underrun information.
-// However, no functionality is currently provided to allow property
-// listeners to trigger user handlers because it is unclear what could
-// be done if a critical stream parameter (buffer size, sample rate,
-// device disconnect) notification arrived. The listeners entail
-// quite a bit of extra code and most likely, a user program wouldn't
-// be prepared for the result anyway. However, we do provide a flag
-// to the client callback function to inform of an over/underrun.
-
-// A structure to hold various information related to the CoreAudio API
-// implementation.
-struct CoreHandle {
- AudioDeviceID id[2]; // device ids
-#if defined( MAC_OS_X_VERSION_10_5 ) && ( MAC_OS_X_VERSION_MIN_REQUIRED >= MAC_OS_X_VERSION_10_5 )
- AudioDeviceIOProcID procId[2];
-#endif
- UInt32 iStream[2]; // device stream index (or first if using multiple)
- UInt32 nStreams[2]; // number of streams to use
- bool xrun[2];
- char *deviceBuffer;
- pthread_cond_t condition;
- int drainCounter; // Tracks callback counts when draining
- bool internalDrain; // Indicates if stop is initiated from callback or not.
-
- CoreHandle()
- :deviceBuffer(0), drainCounter(0), internalDrain(false) { nStreams[0] = 1; nStreams[1] = 1; id[0] = 0; id[1] = 0; xrun[0] = false; xrun[1] = false; }
-};
-
-RtApiCore:: RtApiCore()
-{
-#if defined( AVAILABLE_MAC_OS_X_VERSION_10_6_AND_LATER )
- // This is a largely undocumented but absolutely necessary
- // requirement starting with OS-X 10.6. If not called, queries and
- // updates to various audio device properties are not handled
- // correctly.
- CFRunLoopRef theRunLoop = NULL;
- AudioObjectPropertyAddress property = { kAudioHardwarePropertyRunLoop,
- kAudioObjectPropertyScopeGlobal,
- kAudioObjectPropertyElementMaster };
- OSStatus result = AudioObjectSetPropertyData( kAudioObjectSystemObject, &property, 0, NULL, sizeof(CFRunLoopRef), &theRunLoop);
- if ( result != noErr ) {
- errorText_ = "RtApiCore::RtApiCore: error setting run loop property!";
- error( RtAudioError::WARNING );
- }
-#endif
-}
-
-RtApiCore :: ~RtApiCore()
-{
- // The subclass destructor gets called before the base class
- // destructor, so close an existing stream before deallocating
- // apiDeviceId memory.
- if ( stream_.state != STREAM_CLOSED ) closeStream();
-}
-
-unsigned int RtApiCore :: getDeviceCount( void )
-{
- // Find out how many audio devices there are, if any.
- UInt32 dataSize;
- AudioObjectPropertyAddress propertyAddress = { kAudioHardwarePropertyDevices, kAudioObjectPropertyScopeGlobal, kAudioObjectPropertyElementMaster };
- OSStatus result = AudioObjectGetPropertyDataSize( kAudioObjectSystemObject, &propertyAddress, 0, NULL, &dataSize );
- if ( result != noErr ) {
- errorText_ = "RtApiCore::getDeviceCount: OS-X error getting device info!";
- error( RtAudioError::WARNING );
- return 0;
- }
-
- return dataSize / sizeof( AudioDeviceID );
-}
-
-unsigned int RtApiCore :: getDefaultInputDevice( void )
-{
- unsigned int nDevices = getDeviceCount();
- if ( nDevices <= 1 ) return 0;
-
- AudioDeviceID id;
- UInt32 dataSize = sizeof( AudioDeviceID );
- AudioObjectPropertyAddress property = { kAudioHardwarePropertyDefaultInputDevice, kAudioObjectPropertyScopeGlobal, kAudioObjectPropertyElementMaster };
- OSStatus result = AudioObjectGetPropertyData( kAudioObjectSystemObject, &property, 0, NULL, &dataSize, &id );
- if ( result != noErr ) {
- errorText_ = "RtApiCore::getDefaultInputDevice: OS-X system error getting device.";
- error( RtAudioError::WARNING );
- return 0;
- }
-
- dataSize *= nDevices;
- AudioDeviceID deviceList[ nDevices ];
- property.mSelector = kAudioHardwarePropertyDevices;
- result = AudioObjectGetPropertyData( kAudioObjectSystemObject, &property, 0, NULL, &dataSize, (void *) &deviceList );
- if ( result != noErr ) {
- errorText_ = "RtApiCore::getDefaultInputDevice: OS-X system error getting device IDs.";
- error( RtAudioError::WARNING );
- return 0;
- }
-
- for ( unsigned int i=0; i<nDevices; i++ )
- if ( id == deviceList[i] ) return i;
-
- errorText_ = "RtApiCore::getDefaultInputDevice: No default device found!";
- error( RtAudioError::WARNING );
- return 0;
-}
-
-unsigned int RtApiCore :: getDefaultOutputDevice( void )
-{
- unsigned int nDevices = getDeviceCount();
- if ( nDevices <= 1 ) return 0;
-
- AudioDeviceID id;
- UInt32 dataSize = sizeof( AudioDeviceID );
- AudioObjectPropertyAddress property = { kAudioHardwarePropertyDefaultOutputDevice, kAudioObjectPropertyScopeGlobal, kAudioObjectPropertyElementMaster };
- OSStatus result = AudioObjectGetPropertyData( kAudioObjectSystemObject, &property, 0, NULL, &dataSize, &id );
- if ( result != noErr ) {
- errorText_ = "RtApiCore::getDefaultOutputDevice: OS-X system error getting device.";
- error( RtAudioError::WARNING );
- return 0;
- }
-
- dataSize = sizeof( AudioDeviceID ) * nDevices;
- AudioDeviceID deviceList[ nDevices ];
- property.mSelector = kAudioHardwarePropertyDevices;
- result = AudioObjectGetPropertyData( kAudioObjectSystemObject, &property, 0, NULL, &dataSize, (void *) &deviceList );
- if ( result != noErr ) {
- errorText_ = "RtApiCore::getDefaultOutputDevice: OS-X system error getting device IDs.";
- error( RtAudioError::WARNING );
- return 0;
- }
-
- for ( unsigned int i=0; i<nDevices; i++ )
- if ( id == deviceList[i] ) return i;
-
- errorText_ = "RtApiCore::getDefaultOutputDevice: No default device found!";
- error( RtAudioError::WARNING );
- return 0;
-}
-
-RtAudio::DeviceInfo RtApiCore :: getDeviceInfo( unsigned int device )
-{
- RtAudio::DeviceInfo info;
- info.probed = false;
-
- // Get device ID
- unsigned int nDevices = getDeviceCount();
- if ( nDevices == 0 ) {
- errorText_ = "RtApiCore::getDeviceInfo: no devices found!";
- error( RtAudioError::INVALID_USE );
- return info;
- }
-
- if ( device >= nDevices ) {
- errorText_ = "RtApiCore::getDeviceInfo: device ID is invalid!";
- error( RtAudioError::INVALID_USE );
- return info;
- }
-
- AudioDeviceID deviceList[ nDevices ];
- UInt32 dataSize = sizeof( AudioDeviceID ) * nDevices;
- AudioObjectPropertyAddress property = { kAudioHardwarePropertyDevices,
- kAudioObjectPropertyScopeGlobal,
- kAudioObjectPropertyElementMaster };
- OSStatus result = AudioObjectGetPropertyData( kAudioObjectSystemObject, &property,
- 0, NULL, &dataSize, (void *) &deviceList );
- if ( result != noErr ) {
- errorText_ = "RtApiCore::getDeviceInfo: OS-X system error getting device IDs.";
- error( RtAudioError::WARNING );
- return info;
- }
-
- AudioDeviceID id = deviceList[ device ];
-
- // Get the device name.
- info.name.erase();
- CFStringRef cfname;
- dataSize = sizeof( CFStringRef );
- property.mSelector = kAudioObjectPropertyManufacturer;
- result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, &cfname );
- if ( result != noErr ) {
- errorStream_ << "RtApiCore::probeDeviceInfo: system error (" << getErrorCode( result ) << ") getting device manufacturer.";
- errorText_ = errorStream_.str();
- error( RtAudioError::WARNING );
- return info;
- }
-
- //const char *mname = CFStringGetCStringPtr( cfname, CFStringGetSystemEncoding() );
- int length = CFStringGetLength(cfname);
- char *mname = (char *)malloc(length * 3 + 1);
-#if defined( UNICODE ) || defined( _UNICODE )
- CFStringGetCString(cfname, mname, length * 3 + 1, kCFStringEncodingUTF8);
-#else
- CFStringGetCString(cfname, mname, length * 3 + 1, CFStringGetSystemEncoding());
-#endif
- info.name.append( (const char *)mname, strlen(mname) );
- info.name.append( ": " );
- CFRelease( cfname );
- free(mname);
-
- property.mSelector = kAudioObjectPropertyName;
- result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, &cfname );
- if ( result != noErr ) {
- errorStream_ << "RtApiCore::probeDeviceInfo: system error (" << getErrorCode( result ) << ") getting device name.";
- errorText_ = errorStream_.str();
- error( RtAudioError::WARNING );
- return info;
- }
-
- //const char *name = CFStringGetCStringPtr( cfname, CFStringGetSystemEncoding() );
- length = CFStringGetLength(cfname);
- char *name = (char *)malloc(length * 3 + 1);
-#if defined( UNICODE ) || defined( _UNICODE )
- CFStringGetCString(cfname, name, length * 3 + 1, kCFStringEncodingUTF8);
-#else
- CFStringGetCString(cfname, name, length * 3 + 1, CFStringGetSystemEncoding());
-#endif
- info.name.append( (const char *)name, strlen(name) );
- CFRelease( cfname );
- free(name);
-
- // Get the output stream "configuration".
- AudioBufferList *bufferList = nil;
- property.mSelector = kAudioDevicePropertyStreamConfiguration;
- property.mScope = kAudioDevicePropertyScopeOutput;
- // property.mElement = kAudioObjectPropertyElementWildcard;
- dataSize = 0;
- result = AudioObjectGetPropertyDataSize( id, &property, 0, NULL, &dataSize );
- if ( result != noErr || dataSize == 0 ) {
- errorStream_ << "RtApiCore::getDeviceInfo: system error (" << getErrorCode( result ) << ") getting output stream configuration info for device (" << device << ").";
- errorText_ = errorStream_.str();
- error( RtAudioError::WARNING );
- return info;
- }
-
- // Allocate the AudioBufferList.
- bufferList = (AudioBufferList *) malloc( dataSize );
- if ( bufferList == NULL ) {
- errorText_ = "RtApiCore::getDeviceInfo: memory error allocating output AudioBufferList.";
- error( RtAudioError::WARNING );
- return info;
- }
-
- result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, bufferList );
- if ( result != noErr || dataSize == 0 ) {
- free( bufferList );
- errorStream_ << "RtApiCore::getDeviceInfo: system error (" << getErrorCode( result ) << ") getting output stream configuration for device (" << device << ").";
- errorText_ = errorStream_.str();
- error( RtAudioError::WARNING );
- return info;
- }
-
- // Get output channel information.
- unsigned int i, nStreams = bufferList->mNumberBuffers;
- for ( i=0; i<nStreams; i++ )
- info.outputChannels += bufferList->mBuffers[i].mNumberChannels;
- free( bufferList );
-
- // Get the input stream "configuration".
- property.mScope = kAudioDevicePropertyScopeInput;
- result = AudioObjectGetPropertyDataSize( id, &property, 0, NULL, &dataSize );
- if ( result != noErr || dataSize == 0 ) {
- errorStream_ << "RtApiCore::getDeviceInfo: system error (" << getErrorCode( result ) << ") getting input stream configuration info for device (" << device << ").";
- errorText_ = errorStream_.str();
- error( RtAudioError::WARNING );
- return info;
- }
-
- // Allocate the AudioBufferList.
- bufferList = (AudioBufferList *) malloc( dataSize );
- if ( bufferList == NULL ) {
- errorText_ = "RtApiCore::getDeviceInfo: memory error allocating input AudioBufferList.";
- error( RtAudioError::WARNING );
- return info;
- }
-
- result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, bufferList );
- if (result != noErr || dataSize == 0) {
- free( bufferList );
- errorStream_ << "RtApiCore::getDeviceInfo: system error (" << getErrorCode( result ) << ") getting input stream configuration for device (" << device << ").";
- errorText_ = errorStream_.str();
- error( RtAudioError::WARNING );
- return info;
- }
-
- // Get input channel information.
- nStreams = bufferList->mNumberBuffers;
- for ( i=0; i<nStreams; i++ )
- info.inputChannels += bufferList->mBuffers[i].mNumberChannels;
- free( bufferList );
-
- // If device opens for both playback and capture, we determine the channels.
- if ( info.outputChannels > 0 && info.inputChannels > 0 )
- info.duplexChannels = (info.outputChannels > info.inputChannels) ? info.inputChannels : info.outputChannels;
-
- // Probe the device sample rates.
- bool isInput = false;
- if ( info.outputChannels == 0 ) isInput = true;
-
- // Determine the supported sample rates.
- property.mSelector = kAudioDevicePropertyAvailableNominalSampleRates;
- if ( isInput == false ) property.mScope = kAudioDevicePropertyScopeOutput;
- result = AudioObjectGetPropertyDataSize( id, &property, 0, NULL, &dataSize );
- if ( result != kAudioHardwareNoError || dataSize == 0 ) {
- errorStream_ << "RtApiCore::getDeviceInfo: system error (" << getErrorCode( result ) << ") getting sample rate info.";
- errorText_ = errorStream_.str();
- error( RtAudioError::WARNING );
- return info;
- }
-
- UInt32 nRanges = dataSize / sizeof( AudioValueRange );
- AudioValueRange rangeList[ nRanges ];
- result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, &rangeList );
- if ( result != kAudioHardwareNoError ) {
- errorStream_ << "RtApiCore::getDeviceInfo: system error (" << getErrorCode( result ) << ") getting sample rates.";
- errorText_ = errorStream_.str();
- error( RtAudioError::WARNING );
- return info;
- }
-
- // The sample rate reporting mechanism is a bit of a mystery. It
- // seems that it can either return individual rates or a range of
- // rates. I assume that if the min / max range values are the same,
- // then that represents a single supported rate and if the min / max
- // range values are different, the device supports an arbitrary
- // range of values (though there might be multiple ranges, so we'll
- // use the most conservative range).
- Float64 minimumRate = 1.0, maximumRate = 10000000000.0;
- bool haveValueRange = false;
- info.sampleRates.clear();
- for ( UInt32 i=0; i<nRanges; i++ ) {
- if ( rangeList[i].mMinimum == rangeList[i].mMaximum ) {
- unsigned int tmpSr = (unsigned int) rangeList[i].mMinimum;
- info.sampleRates.push_back( tmpSr );
-
- if ( !info.preferredSampleRate || ( tmpSr <= 48000 && tmpSr > info.preferredSampleRate ) )
- info.preferredSampleRate = tmpSr;
-
- } else {
- haveValueRange = true;
- if ( rangeList[i].mMinimum > minimumRate ) minimumRate = rangeList[i].mMinimum;
- if ( rangeList[i].mMaximum < maximumRate ) maximumRate = rangeList[i].mMaximum;
- }
- }
-
- if ( haveValueRange ) {
- for ( unsigned int k=0; k<MAX_SAMPLE_RATES; k++ ) {
- if ( SAMPLE_RATES[k] >= (unsigned int) minimumRate && SAMPLE_RATES[k] <= (unsigned int) maximumRate ) {
- info.sampleRates.push_back( SAMPLE_RATES[k] );
-
- if ( !info.preferredSampleRate || ( SAMPLE_RATES[k] <= 48000 && SAMPLE_RATES[k] > info.preferredSampleRate ) )
- info.preferredSampleRate = SAMPLE_RATES[k];
- }
- }
- }
-
- // Sort and remove any redundant values
- std::sort( info.sampleRates.begin(), info.sampleRates.end() );
- info.sampleRates.erase( unique( info.sampleRates.begin(), info.sampleRates.end() ), info.sampleRates.end() );
-
- if ( info.sampleRates.size() == 0 ) {
- errorStream_ << "RtApiCore::probeDeviceInfo: No supported sample rates found for device (" << device << ").";
- errorText_ = errorStream_.str();
- error( RtAudioError::WARNING );
- return info;
- }
-
- // CoreAudio always uses 32-bit floating point data for PCM streams.
- // Thus, any other "physical" formats supported by the device are of
- // no interest to the client.
- info.nativeFormats = RTAUDIO_FLOAT32;
-
- if ( info.outputChannels > 0 )
- if ( getDefaultOutputDevice() == device ) info.isDefaultOutput = true;
- if ( info.inputChannels > 0 )
- if ( getDefaultInputDevice() == device ) info.isDefaultInput = true;
-
- info.probed = true;
- return info;
-}
-
-static OSStatus callbackHandler( AudioDeviceID inDevice,
- const AudioTimeStamp* /*inNow*/,
- const AudioBufferList* inInputData,
- const AudioTimeStamp* /*inInputTime*/,
- AudioBufferList* outOutputData,
- const AudioTimeStamp* /*inOutputTime*/,
- void* infoPointer )
-{
- CallbackInfo *info = (CallbackInfo *) infoPointer;
-
- RtApiCore *object = (RtApiCore *) info->object;
- if ( object->callbackEvent( inDevice, inInputData, outOutputData ) == false )
- return kAudioHardwareUnspecifiedError;
- else
- return kAudioHardwareNoError;
-}
-
-static OSStatus xrunListener( AudioObjectID /*inDevice*/,
- UInt32 nAddresses,
- const AudioObjectPropertyAddress properties[],
- void* handlePointer )
-{
- CoreHandle *handle = (CoreHandle *) handlePointer;
- for ( UInt32 i=0; i<nAddresses; i++ ) {
- if ( properties[i].mSelector == kAudioDeviceProcessorOverload ) {
- if ( properties[i].mScope == kAudioDevicePropertyScopeInput )
- handle->xrun[1] = true;
- else
- handle->xrun[0] = true;
- }
- }
-
- return kAudioHardwareNoError;
-}
-
-static OSStatus rateListener( AudioObjectID inDevice,
- UInt32 /*nAddresses*/,
- const AudioObjectPropertyAddress /*properties*/[],
- void* ratePointer )
-{
- Float64 *rate = (Float64 *) ratePointer;
- UInt32 dataSize = sizeof( Float64 );
- AudioObjectPropertyAddress property = { kAudioDevicePropertyNominalSampleRate,
- kAudioObjectPropertyScopeGlobal,
- kAudioObjectPropertyElementMaster };
- AudioObjectGetPropertyData( inDevice, &property, 0, NULL, &dataSize, rate );
- return kAudioHardwareNoError;
-}
-
-bool RtApiCore :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,
- unsigned int firstChannel, unsigned int sampleRate,
- RtAudioFormat format, unsigned int *bufferSize,
- RtAudio::StreamOptions *options )
-{
- // Get device ID
- unsigned int nDevices = getDeviceCount();
- if ( nDevices == 0 ) {
- // This should not happen because a check is made before this function is called.
- errorText_ = "RtApiCore::probeDeviceOpen: no devices found!";
- return FAILURE;
- }
-
- if ( device >= nDevices ) {
- // This should not happen because a check is made before this function is called.
- errorText_ = "RtApiCore::probeDeviceOpen: device ID is invalid!";
- return FAILURE;
- }
-
- AudioDeviceID deviceList[ nDevices ];
- UInt32 dataSize = sizeof( AudioDeviceID ) * nDevices;
- AudioObjectPropertyAddress property = { kAudioHardwarePropertyDevices,
- kAudioObjectPropertyScopeGlobal,
- kAudioObjectPropertyElementMaster };
- OSStatus result = AudioObjectGetPropertyData( kAudioObjectSystemObject, &property,
- 0, NULL, &dataSize, (void *) &deviceList );
- if ( result != noErr ) {
- errorText_ = "RtApiCore::probeDeviceOpen: OS-X system error getting device IDs.";
- return FAILURE;
- }
-
- AudioDeviceID id = deviceList[ device ];
-
- // Setup for stream mode.
- bool isInput = false;
- if ( mode == INPUT ) {
- isInput = true;
- property.mScope = kAudioDevicePropertyScopeInput;
- }
- else
- property.mScope = kAudioDevicePropertyScopeOutput;
-
- // Get the stream "configuration".
- AudioBufferList *bufferList = nil;
- dataSize = 0;
- property.mSelector = kAudioDevicePropertyStreamConfiguration;
- result = AudioObjectGetPropertyDataSize( id, &property, 0, NULL, &dataSize );
- if ( result != noErr || dataSize == 0 ) {
- errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting stream configuration info for device (" << device << ").";
- errorText_ = errorStream_.str();
- return FAILURE;
- }
-
- // Allocate the AudioBufferList.
- bufferList = (AudioBufferList *) malloc( dataSize );
- if ( bufferList == NULL ) {
- errorText_ = "RtApiCore::probeDeviceOpen: memory error allocating AudioBufferList.";
- return FAILURE;
- }
-
- result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, bufferList );
- if (result != noErr || dataSize == 0) {
- free( bufferList );
- errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting stream configuration for device (" << device << ").";
- errorText_ = errorStream_.str();
- return FAILURE;
- }
-
- // Search for one or more streams that contain the desired number of
- // channels. CoreAudio devices can have an arbitrary number of
- // streams and each stream can have an arbitrary number of channels.
- // For each stream, a single buffer of interleaved samples is
- // provided. RtAudio prefers the use of one stream of interleaved
- // data or multiple consecutive single-channel streams. However, we
- // now support multiple consecutive multi-channel streams of
- // interleaved data as well.
- UInt32 iStream, offsetCounter = firstChannel;
- UInt32 nStreams = bufferList->mNumberBuffers;
- bool monoMode = false;
- bool foundStream = false;
-
- // First check that the device supports the requested number of
- // channels.
- UInt32 deviceChannels = 0;
- for ( iStream=0; iStream<nStreams; iStream++ )
- deviceChannels += bufferList->mBuffers[iStream].mNumberChannels;
-
- if ( deviceChannels < ( channels + firstChannel ) ) {
- free( bufferList );
- errorStream_ << "RtApiCore::probeDeviceOpen: the device (" << device << ") does not support the requested channel count.";
- errorText_ = errorStream_.str();
- return FAILURE;
- }
-
- // Look for a single stream meeting our needs.
- UInt32 firstStream, streamCount = 1, streamChannels = 0, channelOffset = 0;
- for ( iStream=0; iStream<nStreams; iStream++ ) {
- streamChannels = bufferList->mBuffers[iStream].mNumberChannels;
- if ( streamChannels >= channels + offsetCounter ) {
- firstStream = iStream;
- channelOffset = offsetCounter;
- foundStream = true;
- break;
- }
- if ( streamChannels > offsetCounter ) break;
- offsetCounter -= streamChannels;
- }
-
- // If we didn't find a single stream above, then we should be able
- // to meet the channel specification with multiple streams.
- if ( foundStream == false ) {
- monoMode = true;
- offsetCounter = firstChannel;
- for ( iStream=0; iStream<nStreams; iStream++ ) {
- streamChannels = bufferList->mBuffers[iStream].mNumberChannels;
- if ( streamChannels > offsetCounter ) break;
- offsetCounter -= streamChannels;
- }
-
- firstStream = iStream;
- channelOffset = offsetCounter;
- Int32 channelCounter = channels + offsetCounter - streamChannels;
-
- if ( streamChannels > 1 ) monoMode = false;
- while ( channelCounter > 0 ) {
- streamChannels = bufferList->mBuffers[++iStream].mNumberChannels;
- if ( streamChannels > 1 ) monoMode = false;
- channelCounter -= streamChannels;
- streamCount++;
- }
- }
-
- free( bufferList );
-
- // Determine the buffer size.
- AudioValueRange bufferRange;
- dataSize = sizeof( AudioValueRange );
- property.mSelector = kAudioDevicePropertyBufferFrameSizeRange;
- result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, &bufferRange );
-
- if ( result != noErr ) {
- errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting buffer size range for device (" << device << ").";
- errorText_ = errorStream_.str();
- return FAILURE;
- }
-
- if ( bufferRange.mMinimum > *bufferSize ) *bufferSize = (unsigned long) bufferRange.mMinimum;
- else if ( bufferRange.mMaximum < *bufferSize ) *bufferSize = (unsigned long) bufferRange.mMaximum;
- if ( options && options->flags & RTAUDIO_MINIMIZE_LATENCY ) *bufferSize = (unsigned long) bufferRange.mMinimum;
-
- // Set the buffer size. For multiple streams, I'm assuming we only
- // need to make this setting for the master channel.
- UInt32 theSize = (UInt32) *bufferSize;
- dataSize = sizeof( UInt32 );
- property.mSelector = kAudioDevicePropertyBufferFrameSize;
- result = AudioObjectSetPropertyData( id, &property, 0, NULL, dataSize, &theSize );
-
- if ( result != noErr ) {
- errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") setting the buffer size for device (" << device << ").";
- errorText_ = errorStream_.str();
- return FAILURE;
- }
-
- // If attempting to setup a duplex stream, the bufferSize parameter
- // MUST be the same in both directions!
- *bufferSize = theSize;
- if ( stream_.mode == OUTPUT && mode == INPUT && *bufferSize != stream_.bufferSize ) {
- errorStream_ << "RtApiCore::probeDeviceOpen: system error setting buffer size for duplex stream on device (" << device << ").";
- errorText_ = errorStream_.str();
- return FAILURE;
- }
-
- stream_.bufferSize = *bufferSize;
- stream_.nBuffers = 1;
-
- // Try to set "hog" mode ... it's not clear to me this is working.
- if ( options && options->flags & RTAUDIO_HOG_DEVICE ) {
- pid_t hog_pid;
- dataSize = sizeof( hog_pid );
- property.mSelector = kAudioDevicePropertyHogMode;
- result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, &hog_pid );
- if ( result != noErr ) {
- errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting 'hog' state!";
- errorText_ = errorStream_.str();
- return FAILURE;
- }
-
- if ( hog_pid != getpid() ) {
- hog_pid = getpid();
- result = AudioObjectSetPropertyData( id, &property, 0, NULL, dataSize, &hog_pid );
- if ( result != noErr ) {
- errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") setting 'hog' state!";
- errorText_ = errorStream_.str();
- return FAILURE;
- }
- }
- }
-
- // Check and if necessary, change the sample rate for the device.
- Float64 nominalRate;
- dataSize = sizeof( Float64 );
- property.mSelector = kAudioDevicePropertyNominalSampleRate;
- result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, &nominalRate );
- if ( result != noErr ) {
- errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting current sample rate.";
- errorText_ = errorStream_.str();
- return FAILURE;
- }
-
- // Only change the sample rate if off by more than 1 Hz.
- if ( fabs( nominalRate - (double)sampleRate ) > 1.0 ) {
-
- // Set a property listener for the sample rate change
- Float64 reportedRate = 0.0;
- AudioObjectPropertyAddress tmp = { kAudioDevicePropertyNominalSampleRate, kAudioObjectPropertyScopeGlobal, kAudioObjectPropertyElementMaster };
- result = AudioObjectAddPropertyListener( id, &tmp, rateListener, (void *) &reportedRate );
- if ( result != noErr ) {
- errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") setting sample rate property listener for device (" << device << ").";
- errorText_ = errorStream_.str();
- return FAILURE;
- }
-
- nominalRate = (Float64) sampleRate;
- result = AudioObjectSetPropertyData( id, &property, 0, NULL, dataSize, &nominalRate );
- if ( result != noErr ) {
- AudioObjectRemovePropertyListener( id, &tmp, rateListener, (void *) &reportedRate );
- errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") setting sample rate for device (" << device << ").";
- errorText_ = errorStream_.str();
- return FAILURE;
- }
-
- // Now wait until the reported nominal rate is what we just set.
- UInt32 microCounter = 0;
- while ( reportedRate != nominalRate ) {
- microCounter += 5000;
- if ( microCounter > 5000000 ) break;
- usleep( 5000 );
- }
-
- // Remove the property listener.
- AudioObjectRemovePropertyListener( id, &tmp, rateListener, (void *) &reportedRate );
-
- if ( microCounter > 5000000 ) {
- errorStream_ << "RtApiCore::probeDeviceOpen: timeout waiting for sample rate update for device (" << device << ").";
- errorText_ = errorStream_.str();
- return FAILURE;
- }
- }
-
- // Now set the stream format for all streams. Also, check the
- // physical format of the device and change that if necessary.
- AudioStreamBasicDescription description;
- dataSize = sizeof( AudioStreamBasicDescription );
- property.mSelector = kAudioStreamPropertyVirtualFormat;
- result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, &description );
- if ( result != noErr ) {
- errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting stream format for device (" << device << ").";
- errorText_ = errorStream_.str();
- return FAILURE;
- }
-
- // Set the sample rate and data format id. However, only make the
- // change if the sample rate is not within 1.0 of the desired
- // rate and the format is not linear pcm.
- bool updateFormat = false;
- if ( fabs( description.mSampleRate - (Float64)sampleRate ) > 1.0 ) {
- description.mSampleRate = (Float64) sampleRate;
- updateFormat = true;
- }
-
- if ( description.mFormatID != kAudioFormatLinearPCM ) {
- description.mFormatID = kAudioFormatLinearPCM;
- updateFormat = true;
- }
-
- if ( updateFormat ) {
- result = AudioObjectSetPropertyData( id, &property, 0, NULL, dataSize, &description );
- if ( result != noErr ) {
- errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") setting sample rate or data format for device (" << device << ").";
- errorText_ = errorStream_.str();
- return FAILURE;
- }
- }
-
- // Now check the physical format.
- property.mSelector = kAudioStreamPropertyPhysicalFormat;
- result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, &description );
- if ( result != noErr ) {
- errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting stream physical format for device (" << device << ").";
- errorText_ = errorStream_.str();
- return FAILURE;
- }
-
- //std::cout << "Current physical stream format:" << std::endl;
- //std::cout << " mBitsPerChan = " << description.mBitsPerChannel << std::endl;
- //std::cout << " aligned high = " << (description.mFormatFlags & kAudioFormatFlagIsAlignedHigh) << ", isPacked = " << (description.mFormatFlags & kAudioFormatFlagIsPacked) << std::endl;
- //std::cout << " bytesPerFrame = " << description.mBytesPerFrame << std::endl;
- //std::cout << " sample rate = " << description.mSampleRate << std::endl;
-
- if ( description.mFormatID != kAudioFormatLinearPCM || description.mBitsPerChannel < 16 ) {
- description.mFormatID = kAudioFormatLinearPCM;
- //description.mSampleRate = (Float64) sampleRate;
- AudioStreamBasicDescription testDescription = description;
- UInt32 formatFlags;
-
- // We'll try higher bit rates first and then work our way down.
- std::vector< std::pair<UInt32, UInt32> > physicalFormats;
- formatFlags = (description.mFormatFlags | kLinearPCMFormatFlagIsFloat) & ~kLinearPCMFormatFlagIsSignedInteger;
- physicalFormats.push_back( std::pair<Float32, UInt32>( 32, formatFlags ) );
- formatFlags = (description.mFormatFlags | kLinearPCMFormatFlagIsSignedInteger | kAudioFormatFlagIsPacked) & ~kLinearPCMFormatFlagIsFloat;
- physicalFormats.push_back( std::pair<Float32, UInt32>( 32, formatFlags ) );
- physicalFormats.push_back( std::pair<Float32, UInt32>( 24, formatFlags ) ); // 24-bit packed
- formatFlags &= ~( kAudioFormatFlagIsPacked | kAudioFormatFlagIsAlignedHigh );
- physicalFormats.push_back( std::pair<Float32, UInt32>( 24.2, formatFlags ) ); // 24-bit in 4 bytes, aligned low
- formatFlags |= kAudioFormatFlagIsAlignedHigh;
- physicalFormats.push_back( std::pair<Float32, UInt32>( 24.4, formatFlags ) ); // 24-bit in 4 bytes, aligned high
- formatFlags = (description.mFormatFlags | kLinearPCMFormatFlagIsSignedInteger | kAudioFormatFlagIsPacked) & ~kLinearPCMFormatFlagIsFloat;
- physicalFormats.push_back( std::pair<Float32, UInt32>( 16, formatFlags ) );
- physicalFormats.push_back( std::pair<Float32, UInt32>( 8, formatFlags ) );
-
- bool setPhysicalFormat = false;
- for( unsigned int i=0; i<physicalFormats.size(); i++ ) {
- testDescription = description;
- testDescription.mBitsPerChannel = (UInt32) physicalFormats[i].first;
- testDescription.mFormatFlags = physicalFormats[i].second;
- if ( (24 == (UInt32)physicalFormats[i].first) && ~( physicalFormats[i].second & kAudioFormatFlagIsPacked ) )
- testDescription.mBytesPerFrame = 4 * testDescription.mChannelsPerFrame;
- else
- testDescription.mBytesPerFrame = testDescription.mBitsPerChannel/8 * testDescription.mChannelsPerFrame;
- testDescription.mBytesPerPacket = testDescription.mBytesPerFrame * testDescription.mFramesPerPacket;
- result = AudioObjectSetPropertyData( id, &property, 0, NULL, dataSize, &testDescription );
- if ( result == noErr ) {
- setPhysicalFormat = true;
- //std::cout << "Updated physical stream format:" << std::endl;
- //std::cout << " mBitsPerChan = " << testDescription.mBitsPerChannel << std::endl;
- //std::cout << " aligned high = " << (testDescription.mFormatFlags & kAudioFormatFlagIsAlignedHigh) << ", isPacked = " << (testDescription.mFormatFlags & kAudioFormatFlagIsPacked) << std::endl;
- //std::cout << " bytesPerFrame = " << testDescription.mBytesPerFrame << std::endl;
- //std::cout << " sample rate = " << testDescription.mSampleRate << std::endl;
- break;
- }
- }
-
- if ( !setPhysicalFormat ) {
- errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") setting physical data format for device (" << device << ").";
- errorText_ = errorStream_.str();
- return FAILURE;
- }
- } // done setting virtual/physical formats.
-
- // Get the stream / device latency.
- UInt32 latency;
- dataSize = sizeof( UInt32 );
- property.mSelector = kAudioDevicePropertyLatency;
- if ( AudioObjectHasProperty( id, &property ) == true ) {
- result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, &latency );
- if ( result == kAudioHardwareNoError ) stream_.latency[ mode ] = latency;
- else {
- errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting device latency for device (" << device << ").";
- errorText_ = errorStream_.str();
- error( RtAudioError::WARNING );
- }
- }
-
- // Byte-swapping: According to AudioHardware.h, the stream data will
- // always be presented in native-endian format, so we should never
- // need to byte swap.
- stream_.doByteSwap[mode] = false;
-
- // From the CoreAudio documentation, PCM data must be supplied as
- // 32-bit floats.
- stream_.userFormat = format;
- stream_.deviceFormat[mode] = RTAUDIO_FLOAT32;
-
- if ( streamCount == 1 )
- stream_.nDeviceChannels[mode] = description.mChannelsPerFrame;
- else // multiple streams
- stream_.nDeviceChannels[mode] = channels;
- stream_.nUserChannels[mode] = channels;
- stream_.channelOffset[mode] = channelOffset; // offset within a CoreAudio stream
- if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) stream_.userInterleaved = false;
- else stream_.userInterleaved = true;
- stream_.deviceInterleaved[mode] = true;
- if ( monoMode == true ) stream_.deviceInterleaved[mode] = false;
-
- // Set flags for buffer conversion.
- stream_.doConvertBuffer[mode] = false;
- if ( stream_.userFormat != stream_.deviceFormat[mode] )
- stream_.doConvertBuffer[mode] = true;
- if ( stream_.nUserChannels[mode] < stream_.nDeviceChannels[mode] )
- stream_.doConvertBuffer[mode] = true;
- if ( streamCount == 1 ) {
- if ( stream_.nUserChannels[mode] > 1 &&
- stream_.userInterleaved != stream_.deviceInterleaved[mode] )
- stream_.doConvertBuffer[mode] = true;
- }
- else if ( monoMode && stream_.userInterleaved )
- stream_.doConvertBuffer[mode] = true;
-
- // Allocate our CoreHandle structure for the stream.
- CoreHandle *handle = 0;
- if ( stream_.apiHandle == 0 ) {
- try {
- handle = new CoreHandle;
- }
- catch ( std::bad_alloc& ) {
- errorText_ = "RtApiCore::probeDeviceOpen: error allocating CoreHandle memory.";
- goto error;
- }
-
- if ( pthread_cond_init( &handle->condition, NULL ) ) {
- errorText_ = "RtApiCore::probeDeviceOpen: error initializing pthread condition variable.";
- goto error;
- }
- stream_.apiHandle = (void *) handle;
- }
- else
- handle = (CoreHandle *) stream_.apiHandle;
- handle->iStream[mode] = firstStream;
- handle->nStreams[mode] = streamCount;
- handle->id[mode] = id;
-
- // Allocate necessary internal buffers.
- unsigned long bufferBytes;
- bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat );
- // stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 );
- stream_.userBuffer[mode] = (char *) malloc( bufferBytes * sizeof(char) );
- memset( stream_.userBuffer[mode], 0, bufferBytes * sizeof(char) );
- if ( stream_.userBuffer[mode] == NULL ) {
- errorText_ = "RtApiCore::probeDeviceOpen: error allocating user buffer memory.";
- goto error;
- }
-
- // If possible, we will make use of the CoreAudio stream buffers as
- // "device buffers". However, we can't do this if using multiple
- // streams.
- if ( stream_.doConvertBuffer[mode] && handle->nStreams[mode] > 1 ) {
-
- bool makeBuffer = true;
- bufferBytes = stream_.nDeviceChannels[mode] * formatBytes( stream_.deviceFormat[mode] );
- if ( mode == INPUT ) {
- if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) {
- unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] );
- if ( bufferBytes <= bytesOut ) makeBuffer = false;
- }
- }
-
- if ( makeBuffer ) {
- bufferBytes *= *bufferSize;
- if ( stream_.deviceBuffer ) free( stream_.deviceBuffer );
- stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 );
- if ( stream_.deviceBuffer == NULL ) {
- errorText_ = "RtApiCore::probeDeviceOpen: error allocating device buffer memory.";
- goto error;
- }
- }
- }
-
- stream_.sampleRate = sampleRate;
- stream_.device[mode] = device;
- stream_.state = STREAM_STOPPED;
- stream_.callbackInfo.object = (void *) this;
-
- // Setup the buffer conversion information structure.
- if ( stream_.doConvertBuffer[mode] ) {
- if ( streamCount > 1 ) setConvertInfo( mode, 0 );
- else setConvertInfo( mode, channelOffset );
- }
-
- if ( mode == INPUT && stream_.mode == OUTPUT && stream_.device[0] == device )
- // Only one callback procedure per device.
- stream_.mode = DUPLEX;
- else {
-#if defined( MAC_OS_X_VERSION_10_5 ) && ( MAC_OS_X_VERSION_MIN_REQUIRED >= MAC_OS_X_VERSION_10_5 )
- result = AudioDeviceCreateIOProcID( id, callbackHandler, (void *) &stream_.callbackInfo, &handle->procId[mode] );
-#else
- // deprecated in favor of AudioDeviceCreateIOProcID()
- result = AudioDeviceAddIOProc( id, callbackHandler, (void *) &stream_.callbackInfo );
-#endif
- if ( result != noErr ) {
- errorStream_ << "RtApiCore::probeDeviceOpen: system error setting callback for device (" << device << ").";
- errorText_ = errorStream_.str();
- goto error;
- }
- if ( stream_.mode == OUTPUT && mode == INPUT )
- stream_.mode = DUPLEX;
- else
- stream_.mode = mode;
- }
-
- // Setup the device property listener for over/underload.
- property.mSelector = kAudioDeviceProcessorOverload;
- property.mScope = kAudioObjectPropertyScopeGlobal;
- result = AudioObjectAddPropertyListener( id, &property, xrunListener, (void *) handle );
-
- return SUCCESS;
-
- error:
- if ( handle ) {
- pthread_cond_destroy( &handle->condition );
- delete handle;
- stream_.apiHandle = 0;
- }
-
- for ( int i=0; i<2; i++ ) {
- if ( stream_.userBuffer[i] ) {
- free( stream_.userBuffer[i] );
- stream_.userBuffer[i] = 0;
- }
- }
-
- if ( stream_.deviceBuffer ) {
- free( stream_.deviceBuffer );
- stream_.deviceBuffer = 0;
- }
-
- stream_.state = STREAM_CLOSED;
- return FAILURE;
-}
-
-void RtApiCore :: closeStream( void )
-{
- if ( stream_.state == STREAM_CLOSED ) {
- errorText_ = "RtApiCore::closeStream(): no open stream to close!";
- error( RtAudioError::WARNING );
- return;
- }
-
- CoreHandle *handle = (CoreHandle *) stream_.apiHandle;
- if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
- if (handle) {
- AudioObjectPropertyAddress property = { kAudioHardwarePropertyDevices,
- kAudioObjectPropertyScopeGlobal,
- kAudioObjectPropertyElementMaster };
-
- property.mSelector = kAudioDeviceProcessorOverload;
- property.mScope = kAudioObjectPropertyScopeGlobal;
- if (AudioObjectRemovePropertyListener( handle->id[0], &property, xrunListener, (void *) handle ) != noErr) {
- errorText_ = "RtApiCore::closeStream(): error removing property listener!";
- error( RtAudioError::WARNING );
- }
- }
- if ( stream_.state == STREAM_RUNNING )
- AudioDeviceStop( handle->id[0], callbackHandler );
-#if defined( MAC_OS_X_VERSION_10_5 ) && ( MAC_OS_X_VERSION_MIN_REQUIRED >= MAC_OS_X_VERSION_10_5 )
- AudioDeviceDestroyIOProcID( handle->id[0], handle->procId[0] );
-#else
- // deprecated in favor of AudioDeviceDestroyIOProcID()
- AudioDeviceRemoveIOProc( handle->id[0], callbackHandler );
-#endif
- }
-
- if ( stream_.mode == INPUT || ( stream_.mode == DUPLEX && stream_.device[0] != stream_.device[1] ) ) {
- if (handle) {
- AudioObjectPropertyAddress property = { kAudioHardwarePropertyDevices,
- kAudioObjectPropertyScopeGlobal,
- kAudioObjectPropertyElementMaster };
-
- property.mSelector = kAudioDeviceProcessorOverload;
- property.mScope = kAudioObjectPropertyScopeGlobal;
- if (AudioObjectRemovePropertyListener( handle->id[1], &property, xrunListener, (void *) handle ) != noErr) {
- errorText_ = "RtApiCore::closeStream(): error removing property listener!";
- error( RtAudioError::WARNING );
- }
- }
- if ( stream_.state == STREAM_RUNNING )
- AudioDeviceStop( handle->id[1], callbackHandler );
-#if defined( MAC_OS_X_VERSION_10_5 ) && ( MAC_OS_X_VERSION_MIN_REQUIRED >= MAC_OS_X_VERSION_10_5 )
- AudioDeviceDestroyIOProcID( handle->id[1], handle->procId[1] );
-#else
- // deprecated in favor of AudioDeviceDestroyIOProcID()
- AudioDeviceRemoveIOProc( handle->id[1], callbackHandler );
-#endif
- }
-
- for ( int i=0; i<2; i++ ) {
- if ( stream_.userBuffer[i] ) {
- free( stream_.userBuffer[i] );
- stream_.userBuffer[i] = 0;
- }
- }
-
- if ( stream_.deviceBuffer ) {
- free( stream_.deviceBuffer );
- stream_.deviceBuffer = 0;
- }
-
- // Destroy pthread condition variable.
- pthread_cond_destroy( &handle->condition );
- delete handle;
- stream_.apiHandle = 0;
-
- stream_.mode = UNINITIALIZED;
- stream_.state = STREAM_CLOSED;
-}
-
-void RtApiCore :: startStream( void )
-{
- verifyStream();
- if ( stream_.state == STREAM_RUNNING ) {
- errorText_ = "RtApiCore::startStream(): the stream is already running!";
- error( RtAudioError::WARNING );
- return;
- }
-
- OSStatus result = noErr;
- CoreHandle *handle = (CoreHandle *) stream_.apiHandle;
- if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
-
- result = AudioDeviceStart( handle->id[0], callbackHandler );
- if ( result != noErr ) {
- errorStream_ << "RtApiCore::startStream: system error (" << getErrorCode( result ) << ") starting callback procedure on device (" << stream_.device[0] << ").";
- errorText_ = errorStream_.str();
- goto unlock;
- }
- }
-
- if ( stream_.mode == INPUT ||
- ( stream_.mode == DUPLEX && stream_.device[0] != stream_.device[1] ) ) {
-
- result = AudioDeviceStart( handle->id[1], callbackHandler );
- if ( result != noErr ) {
- errorStream_ << "RtApiCore::startStream: system error starting input callback procedure on device (" << stream_.device[1] << ").";
- errorText_ = errorStream_.str();
- goto unlock;
- }
- }
-
- handle->drainCounter = 0;
- handle->internalDrain = false;
- stream_.state = STREAM_RUNNING;
-
- unlock:
- if ( result == noErr ) return;
- error( RtAudioError::SYSTEM_ERROR );
-}
-
-void RtApiCore :: stopStream( void )
-{
- verifyStream();
- if ( stream_.state == STREAM_STOPPED ) {
- errorText_ = "RtApiCore::stopStream(): the stream is already stopped!";
- error( RtAudioError::WARNING );
- return;
- }
-
- OSStatus result = noErr;
- CoreHandle *handle = (CoreHandle *) stream_.apiHandle;
- if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
-
- if ( handle->drainCounter == 0 ) {
- handle->drainCounter = 2;
- pthread_cond_wait( &handle->condition, &stream_.mutex ); // block until signaled
- }
-
- result = AudioDeviceStop( handle->id[0], callbackHandler );
- if ( result != noErr ) {
- errorStream_ << "RtApiCore::stopStream: system error (" << getErrorCode( result ) << ") stopping callback procedure on device (" << stream_.device[0] << ").";
- errorText_ = errorStream_.str();
- goto unlock;
- }
- }
-
- if ( stream_.mode == INPUT || ( stream_.mode == DUPLEX && stream_.device[0] != stream_.device[1] ) ) {
-
- result = AudioDeviceStop( handle->id[1], callbackHandler );
- if ( result != noErr ) {
- errorStream_ << "RtApiCore::stopStream: system error (" << getErrorCode( result ) << ") stopping input callback procedure on device (" << stream_.device[1] << ").";
- errorText_ = errorStream_.str();
- goto unlock;
- }
- }
-
- stream_.state = STREAM_STOPPED;
-
- unlock:
- if ( result == noErr ) return;
- error( RtAudioError::SYSTEM_ERROR );
-}
-
-void RtApiCore :: abortStream( void )
-{
- verifyStream();
- if ( stream_.state == STREAM_STOPPED ) {
- errorText_ = "RtApiCore::abortStream(): the stream is already stopped!";
- error( RtAudioError::WARNING );
- return;
- }
-
- CoreHandle *handle = (CoreHandle *) stream_.apiHandle;
- handle->drainCounter = 2;
-
- stopStream();
-}
-
-// This function will be called by a spawned thread when the user
-// callback function signals that the stream should be stopped or
-// aborted. It is better to handle it this way because the
-// callbackEvent() function probably should return before the AudioDeviceStop()
-// function is called.
-static void *coreStopStream( void *ptr )
-{
- CallbackInfo *info = (CallbackInfo *) ptr;
- RtApiCore *object = (RtApiCore *) info->object;
-
- object->stopStream();
- pthread_exit( NULL );
-}
-
-bool RtApiCore :: callbackEvent( AudioDeviceID deviceId,
- const AudioBufferList *inBufferList,
- const AudioBufferList *outBufferList )
-{
- if ( stream_.state == STREAM_STOPPED || stream_.state == STREAM_STOPPING ) return SUCCESS;
- if ( stream_.state == STREAM_CLOSED ) {
- errorText_ = "RtApiCore::callbackEvent(): the stream is closed ... this shouldn't happen!";
- error( RtAudioError::WARNING );
- return FAILURE;
- }
-
- CallbackInfo *info = (CallbackInfo *) &stream_.callbackInfo;
- CoreHandle *handle = (CoreHandle *) stream_.apiHandle;
-
- // Check if we were draining the stream and signal is finished.
- if ( handle->drainCounter > 3 ) {
- ThreadHandle threadId;
-
- stream_.state = STREAM_STOPPING;
- if ( handle->internalDrain == true )
- pthread_create( &threadId, NULL, coreStopStream, info );
- else // external call to stopStream()
- pthread_cond_signal( &handle->condition );
- return SUCCESS;
- }
-
- AudioDeviceID outputDevice = handle->id[0];
-
- // Invoke user callback to get fresh output data UNLESS we are
- // draining stream or duplex mode AND the input/output devices are
- // different AND this function is called for the input device.
- if ( handle->drainCounter == 0 && ( stream_.mode != DUPLEX || deviceId == outputDevice ) ) {
- RtAudioCallback callback = (RtAudioCallback) info->callback;
- double streamTime = getStreamTime();
- RtAudioStreamStatus status = 0;
- if ( stream_.mode != INPUT && handle->xrun[0] == true ) {
- status |= RTAUDIO_OUTPUT_UNDERFLOW;
- handle->xrun[0] = false;
- }
- if ( stream_.mode != OUTPUT && handle->xrun[1] == true ) {
- status |= RTAUDIO_INPUT_OVERFLOW;
- handle->xrun[1] = false;
- }
-
- int cbReturnValue = callback( stream_.userBuffer[0], stream_.userBuffer[1],
- stream_.bufferSize, streamTime, status, info->userData );
- if ( cbReturnValue == 2 ) {
- stream_.state = STREAM_STOPPING;
- handle->drainCounter = 2;
- abortStream();
- return SUCCESS;
- }
- else if ( cbReturnValue == 1 ) {
- handle->drainCounter = 1;
- handle->internalDrain = true;
- }
- }
-
- if ( stream_.mode == OUTPUT || ( stream_.mode == DUPLEX && deviceId == outputDevice ) ) {
-
- if ( handle->drainCounter > 1 ) { // write zeros to the output stream
-
- if ( handle->nStreams[0] == 1 ) {
- memset( outBufferList->mBuffers[handle->iStream[0]].mData,
- 0,
- outBufferList->mBuffers[handle->iStream[0]].mDataByteSize );
- }
- else { // fill multiple streams with zeros
- for ( unsigned int i=0; i<handle->nStreams[0]; i++ ) {
- memset( outBufferList->mBuffers[handle->iStream[0]+i].mData,
- 0,
- outBufferList->mBuffers[handle->iStream[0]+i].mDataByteSize );
- }
- }
- }
- else if ( handle->nStreams[0] == 1 ) {
- if ( stream_.doConvertBuffer[0] ) { // convert directly to CoreAudio stream buffer
- convertBuffer( (char *) outBufferList->mBuffers[handle->iStream[0]].mData,
- stream_.userBuffer[0], stream_.convertInfo[0] );
- }
- else { // copy from user buffer
- memcpy( outBufferList->mBuffers[handle->iStream[0]].mData,
- stream_.userBuffer[0],
- outBufferList->mBuffers[handle->iStream[0]].mDataByteSize );
- }
- }
- else { // fill multiple streams
- Float32 *inBuffer = (Float32 *) stream_.userBuffer[0];
- if ( stream_.doConvertBuffer[0] ) {
- convertBuffer( stream_.deviceBuffer, stream_.userBuffer[0], stream_.convertInfo[0] );
- inBuffer = (Float32 *) stream_.deviceBuffer;
- }
-
- if ( stream_.deviceInterleaved[0] == false ) { // mono mode
- UInt32 bufferBytes = outBufferList->mBuffers[handle->iStream[0]].mDataByteSize;
- for ( unsigned int i=0; i<stream_.nUserChannels[0]; i++ ) {
- memcpy( outBufferList->mBuffers[handle->iStream[0]+i].mData,
- (void *)&inBuffer[i*stream_.bufferSize], bufferBytes );
- }
- }
- else { // fill multiple multi-channel streams with interleaved data
- UInt32 streamChannels, channelsLeft, inJump, outJump, inOffset;
- Float32 *out, *in;
-
- bool inInterleaved = ( stream_.userInterleaved ) ? true : false;
- UInt32 inChannels = stream_.nUserChannels[0];
- if ( stream_.doConvertBuffer[0] ) {
- inInterleaved = true; // device buffer will always be interleaved for nStreams > 1 and not mono mode
- inChannels = stream_.nDeviceChannels[0];
- }
-
- if ( inInterleaved ) inOffset = 1;
- else inOffset = stream_.bufferSize;
-
- channelsLeft = inChannels;
- for ( unsigned int i=0; i<handle->nStreams[0]; i++ ) {
- in = inBuffer;
- out = (Float32 *) outBufferList->mBuffers[handle->iStream[0]+i].mData;
- streamChannels = outBufferList->mBuffers[handle->iStream[0]+i].mNumberChannels;
-
- outJump = 0;
- // Account for possible channel offset in first stream
- if ( i == 0 && stream_.channelOffset[0] > 0 ) {
- streamChannels -= stream_.channelOffset[0];
- outJump = stream_.channelOffset[0];
- out += outJump;
- }
-
- // Account for possible unfilled channels at end of the last stream
- if ( streamChannels > channelsLeft ) {
- outJump = streamChannels - channelsLeft;
- streamChannels = channelsLeft;
- }
-
- // Determine input buffer offsets and skips
- if ( inInterleaved ) {
- inJump = inChannels;
- in += inChannels - channelsLeft;
- }
- else {
- inJump = 1;
- in += (inChannels - channelsLeft) * inOffset;
- }
-
- for ( unsigned int i=0; i<stream_.bufferSize; i++ ) {
- for ( unsigned int j=0; j<streamChannels; j++ ) {
- *out++ = in[j*inOffset];
- }
- out += outJump;
- in += inJump;
- }
- channelsLeft -= streamChannels;
- }
- }
- }
- }
-
- // Don't bother draining input
- if ( handle->drainCounter ) {
- handle->drainCounter++;
- goto unlock;
- }
-
- AudioDeviceID inputDevice;
- inputDevice = handle->id[1];
- if ( stream_.mode == INPUT || ( stream_.mode == DUPLEX && deviceId == inputDevice ) ) {
-
- if ( handle->nStreams[1] == 1 ) {
- if ( stream_.doConvertBuffer[1] ) { // convert directly from CoreAudio stream buffer
- convertBuffer( stream_.userBuffer[1],
- (char *) inBufferList->mBuffers[handle->iStream[1]].mData,
- stream_.convertInfo[1] );
- }
- else { // copy to user buffer
- memcpy( stream_.userBuffer[1],
- inBufferList->mBuffers[handle->iStream[1]].mData,
- inBufferList->mBuffers[handle->iStream[1]].mDataByteSize );
- }
- }
- else { // read from multiple streams
- Float32 *outBuffer = (Float32 *) stream_.userBuffer[1];
- if ( stream_.doConvertBuffer[1] ) outBuffer = (Float32 *) stream_.deviceBuffer;
-
- if ( stream_.deviceInterleaved[1] == false ) { // mono mode
- UInt32 bufferBytes = inBufferList->mBuffers[handle->iStream[1]].mDataByteSize;
- for ( unsigned int i=0; i<stream_.nUserChannels[1]; i++ ) {
- memcpy( (void *)&outBuffer[i*stream_.bufferSize],
- inBufferList->mBuffers[handle->iStream[1]+i].mData, bufferBytes );
- }
- }
- else { // read from multiple multi-channel streams
- UInt32 streamChannels, channelsLeft, inJump, outJump, outOffset;
- Float32 *out, *in;
-
- bool outInterleaved = ( stream_.userInterleaved ) ? true : false;
- UInt32 outChannels = stream_.nUserChannels[1];
- if ( stream_.doConvertBuffer[1] ) {
- outInterleaved = true; // device buffer will always be interleaved for nStreams > 1 and not mono mode
- outChannels = stream_.nDeviceChannels[1];
- }
-
- if ( outInterleaved ) outOffset = 1;
- else outOffset = stream_.bufferSize;
-
- channelsLeft = outChannels;
- for ( unsigned int i=0; i<handle->nStreams[1]; i++ ) {
- out = outBuffer;
- in = (Float32 *) inBufferList->mBuffers[handle->iStream[1]+i].mData;
- streamChannels = inBufferList->mBuffers[handle->iStream[1]+i].mNumberChannels;
-
- inJump = 0;
- // Account for possible channel offset in first stream
- if ( i == 0 && stream_.channelOffset[1] > 0 ) {
- streamChannels -= stream_.channelOffset[1];
- inJump = stream_.channelOffset[1];
- in += inJump;
- }
-
- // Account for possible unread channels at end of the last stream
- if ( streamChannels > channelsLeft ) {
- inJump = streamChannels - channelsLeft;
- streamChannels = channelsLeft;
- }
-
- // Determine output buffer offsets and skips
- if ( outInterleaved ) {
- outJump = outChannels;
- out += outChannels - channelsLeft;
- }
- else {
- outJump = 1;
- out += (outChannels - channelsLeft) * outOffset;
- }
-
- for ( unsigned int i=0; i<stream_.bufferSize; i++ ) {
- for ( unsigned int j=0; j<streamChannels; j++ ) {
- out[j*outOffset] = *in++;
- }
- out += outJump;
- in += inJump;
- }
- channelsLeft -= streamChannels;
- }
- }
-
- if ( stream_.doConvertBuffer[1] ) { // convert from our internal "device" buffer
- convertBuffer( stream_.userBuffer[1],
- stream_.deviceBuffer,
- stream_.convertInfo[1] );
- }
- }
- }
-
- unlock:
- //MUTEX_UNLOCK( &stream_.mutex );
-
- RtApi::tickStreamTime();
- return SUCCESS;
-}
-
-const char* RtApiCore :: getErrorCode( OSStatus code )
-{
- switch( code ) {
-
- case kAudioHardwareNotRunningError:
- return "kAudioHardwareNotRunningError";
-
- case kAudioHardwareUnspecifiedError:
- return "kAudioHardwareUnspecifiedError";
-
- case kAudioHardwareUnknownPropertyError:
- return "kAudioHardwareUnknownPropertyError";
-
- case kAudioHardwareBadPropertySizeError:
- return "kAudioHardwareBadPropertySizeError";
-
- case kAudioHardwareIllegalOperationError:
- return "kAudioHardwareIllegalOperationError";
-
- case kAudioHardwareBadObjectError:
- return "kAudioHardwareBadObjectError";
-
- case kAudioHardwareBadDeviceError:
- return "kAudioHardwareBadDeviceError";
-
- case kAudioHardwareBadStreamError:
- return "kAudioHardwareBadStreamError";
-
- case kAudioHardwareUnsupportedOperationError:
- return "kAudioHardwareUnsupportedOperationError";
-
- case kAudioDeviceUnsupportedFormatError:
- return "kAudioDeviceUnsupportedFormatError";
-
- case kAudioDevicePermissionsError:
- return "kAudioDevicePermissionsError";
-
- default:
- return "CoreAudio unknown error";
- }
-}
-
- //******************** End of __MACOSX_CORE__ *********************//
-#endif
-
-#if defined(__UNIX_JACK__)
-
-// JACK is a low-latency audio server, originally written for the
-// GNU/Linux operating system and now also ported to OS-X. It can
-// connect a number of different applications to an audio device, as
-// well as allowing them to share audio between themselves.
-//
-// When using JACK with RtAudio, "devices" refer to JACK clients that
-// have ports connected to the server. The JACK server is typically
-// started in a terminal as follows:
-//
-// .jackd -d alsa -d hw:0
-//
-// or through an interface program such as qjackctl. Many of the
-// parameters normally set for a stream are fixed by the JACK server
-// and can be specified when the JACK server is started. In
-// particular,
-//
-// .jackd -d alsa -d hw:0 -r 44100 -p 512 -n 4
-//
-// specifies a sample rate of 44100 Hz, a buffer size of 512 sample
-// frames, and number of buffers = 4. Once the server is running, it
-// is not possible to override these values. If the values are not
-// specified in the command-line, the JACK server uses default values.
-//
-// The JACK server does not have to be running when an instance of
-// RtApiJack is created, though the function getDeviceCount() will
-// report 0 devices found until JACK has been started. When no
-// devices are available (i.e., the JACK server is not running), a
-// stream cannot be opened.
-
-#include <jack/jack.h>
-#include <unistd.h>
-#include <cstdio>
-
-// A structure to hold various information related to the Jack API
-// implementation.
-struct JackHandle {
- jack_client_t *client;
- jack_port_t **ports[2];
- std::string deviceName[2];
- bool xrun[2];
- pthread_cond_t condition;
- int drainCounter; // Tracks callback counts when draining
- bool internalDrain; // Indicates if stop is initiated from callback or not.
-
- JackHandle()
- :client(0), drainCounter(0), internalDrain(false) { ports[0] = 0; ports[1] = 0; xrun[0] = false; xrun[1] = false; }
-};
-
-static void jackSilentError( const char * ) {};
-
-RtApiJack :: RtApiJack()
-{
- // Nothing to do here.
-#if !defined(__RTAUDIO_DEBUG__)
- // Turn off Jack's internal error reporting.
- jack_set_error_function( &jackSilentError );
-#endif
-}
-
-RtApiJack :: ~RtApiJack()
-{
- if ( stream_.state != STREAM_CLOSED ) closeStream();
-}
-
-unsigned int RtApiJack :: getDeviceCount( void )
-{
- // See if we can become a jack client.
- jack_options_t options = (jack_options_t) ( JackNoStartServer ); //JackNullOption;
- jack_status_t *status = NULL;
- jack_client_t *client = jack_client_open( "RtApiJackCount", options, status );
- if ( client == 0 ) return 0;
-
- const char **ports;
- std::string port, previousPort;
- unsigned int nChannels = 0, nDevices = 0;
- ports = jack_get_ports( client, NULL, NULL, 0 );
- if ( ports ) {
- // Parse the port names up to the first colon (:).
- size_t iColon = 0;
- do {
- port = (char *) ports[ nChannels ];
- iColon = port.find(":");
- if ( iColon != std::string::npos ) {
- port = port.substr( 0, iColon + 1 );
- if ( port != previousPort ) {
- nDevices++;
- previousPort = port;
- }
- }
- } while ( ports[++nChannels] );
- free( ports );
- }
-
- jack_client_close( client );
- return nDevices;
-}
-
-RtAudio::DeviceInfo RtApiJack :: getDeviceInfo( unsigned int device )
-{
- RtAudio::DeviceInfo info;
- info.probed = false;
-
- jack_options_t options = (jack_options_t) ( JackNoStartServer ); //JackNullOption
- jack_status_t *status = NULL;
- jack_client_t *client = jack_client_open( "RtApiJackInfo", options, status );
- if ( client == 0 ) {
- errorText_ = "RtApiJack::getDeviceInfo: Jack server not found or connection error!";
- error( RtAudioError::WARNING );
- return info;
- }
-
- const char **ports;
- std::string port, previousPort;
- unsigned int nPorts = 0, nDevices = 0;
- ports = jack_get_ports( client, NULL, NULL, 0 );
- if ( ports ) {
- // Parse the port names up to the first colon (:).
- size_t iColon = 0;
- do {
- port = (char *) ports[ nPorts ];
- iColon = port.find(":");
- if ( iColon != std::string::npos ) {
- port = port.substr( 0, iColon );
- if ( port != previousPort ) {
- if ( nDevices == device ) info.name = port;
- nDevices++;
- previousPort = port;
- }
- }
- } while ( ports[++nPorts] );
- free( ports );
- }
-
- if ( device >= nDevices ) {
- jack_client_close( client );
- errorText_ = "RtApiJack::getDeviceInfo: device ID is invalid!";
- error( RtAudioError::INVALID_USE );
- return info;
- }
-
- // Get the current jack server sample rate.
- info.sampleRates.clear();
-
- info.preferredSampleRate = jack_get_sample_rate( client );
- info.sampleRates.push_back( info.preferredSampleRate );
-
- // Count the available ports containing the client name as device
- // channels. Jack "input ports" equal RtAudio output channels.
- unsigned int nChannels = 0;
- ports = jack_get_ports( client, info.name.c_str(), NULL, JackPortIsInput );
- if ( ports ) {
- while ( ports[ nChannels ] ) nChannels++;
- free( ports );
- info.outputChannels = nChannels;
- }
-
- // Jack "output ports" equal RtAudio input channels.
- nChannels = 0;
- ports = jack_get_ports( client, info.name.c_str(), NULL, JackPortIsOutput );
- if ( ports ) {
- while ( ports[ nChannels ] ) nChannels++;
- free( ports );
- info.inputChannels = nChannels;
- }
-
- if ( info.outputChannels == 0 && info.inputChannels == 0 ) {
- jack_client_close(client);
- errorText_ = "RtApiJack::getDeviceInfo: error determining Jack input/output channels!";
- error( RtAudioError::WARNING );
- return info;
- }
-
- // If device opens for both playback and capture, we determine the channels.
- if ( info.outputChannels > 0 && info.inputChannels > 0 )
- info.duplexChannels = (info.outputChannels > info.inputChannels) ? info.inputChannels : info.outputChannels;
-
- // Jack always uses 32-bit floats.
- info.nativeFormats = RTAUDIO_FLOAT32;
-
- // Jack doesn't provide default devices so we'll use the first available one.
- if ( device == 0 && info.outputChannels > 0 )
- info.isDefaultOutput = true;
- if ( device == 0 && info.inputChannels > 0 )
- info.isDefaultInput = true;
-
- jack_client_close(client);
- info.probed = true;
- return info;
-}
-
-static int jackCallbackHandler( jack_nframes_t nframes, void *infoPointer )
-{
- CallbackInfo *info = (CallbackInfo *) infoPointer;
-
- RtApiJack *object = (RtApiJack *) info->object;
- if ( object->callbackEvent( (unsigned long) nframes ) == false ) return 1;
-
- return 0;
-}
-
-// This function will be called by a spawned thread when the Jack
-// server signals that it is shutting down. It is necessary to handle
-// it this way because the jackShutdown() function must return before
-// the jack_deactivate() function (in closeStream()) will return.
-static void *jackCloseStream( void *ptr )
-{
- CallbackInfo *info = (CallbackInfo *) ptr;
- RtApiJack *object = (RtApiJack *) info->object;
-
- object->closeStream();
-
- pthread_exit( NULL );
-}
-static void jackShutdown( void *infoPointer )
-{
- CallbackInfo *info = (CallbackInfo *) infoPointer;
- RtApiJack *object = (RtApiJack *) info->object;
-
- // Check current stream state. If stopped, then we'll assume this
- // was called as a result of a call to RtApiJack::stopStream (the
- // deactivation of a client handle causes this function to be called).
- // If not, we'll assume the Jack server is shutting down or some
- // other problem occurred and we should close the stream.
- if ( object->isStreamRunning() == false ) return;
-
- ThreadHandle threadId;
- pthread_create( &threadId, NULL, jackCloseStream, info );
- std::cerr << "\nRtApiJack: the Jack server is shutting down this client ... stream stopped and closed!!\n" << std::endl;
-}
-
-static int jackXrun( void *infoPointer )
-{
- JackHandle *handle = (JackHandle *) infoPointer;
-
- if ( handle->ports[0] ) handle->xrun[0] = true;
- if ( handle->ports[1] ) handle->xrun[1] = true;
-
- return 0;
-}
-
-bool RtApiJack :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,
- unsigned int firstChannel, unsigned int sampleRate,
- RtAudioFormat format, unsigned int *bufferSize,
- RtAudio::StreamOptions *options )
-{
- JackHandle *handle = (JackHandle *) stream_.apiHandle;
-
- // Look for jack server and try to become a client (only do once per stream).
- jack_client_t *client = 0;
- if ( mode == OUTPUT || ( mode == INPUT && stream_.mode != OUTPUT ) ) {
- jack_options_t jackoptions = (jack_options_t) ( JackNoStartServer ); //JackNullOption;
- jack_status_t *status = NULL;
- if ( options && !options->streamName.empty() )
- client = jack_client_open( options->streamName.c_str(), jackoptions, status );
- else
- client = jack_client_open( "RtApiJack", jackoptions, status );
- if ( client == 0 ) {
- errorText_ = "RtApiJack::probeDeviceOpen: Jack server not found or connection error!";
- error( RtAudioError::WARNING );
- return FAILURE;
- }
- }
- else {
- // The handle must have been created on an earlier pass.
- client = handle->client;
- }
-
- const char **ports;
- std::string port, previousPort, deviceName;
- unsigned int nPorts = 0, nDevices = 0;
- ports = jack_get_ports( client, NULL, NULL, 0 );
- if ( ports ) {
- // Parse the port names up to the first colon (:).
- size_t iColon = 0;
- do {
- port = (char *) ports[ nPorts ];
- iColon = port.find(":");
- if ( iColon != std::string::npos ) {
- port = port.substr( 0, iColon );
- if ( port != previousPort ) {
- if ( nDevices == device ) deviceName = port;
- nDevices++;
- previousPort = port;
- }
- }
- } while ( ports[++nPorts] );
- free( ports );
- }
-
- if ( device >= nDevices ) {
- errorText_ = "RtApiJack::probeDeviceOpen: device ID is invalid!";
- return FAILURE;
- }
-
- // Count the available ports containing the client name as device
- // channels. Jack "input ports" equal RtAudio output channels.
- unsigned int nChannels = 0;
- unsigned long flag = JackPortIsInput;
- if ( mode == INPUT ) flag = JackPortIsOutput;
- ports = jack_get_ports( client, deviceName.c_str(), NULL, flag );
- if ( ports ) {
- while ( ports[ nChannels ] ) nChannels++;
- free( ports );
- }
-
- // Compare the jack ports for specified client to the requested number of channels.
- if ( nChannels < (channels + firstChannel) ) {
- errorStream_ << "RtApiJack::probeDeviceOpen: requested number of channels (" << channels << ") + offset (" << firstChannel << ") not found for specified device (" << device << ":" << deviceName << ").";
- errorText_ = errorStream_.str();
- return FAILURE;
- }
-
- // Check the jack server sample rate.
- unsigned int jackRate = jack_get_sample_rate( client );
- if ( sampleRate != jackRate ) {
- jack_client_close( client );
- errorStream_ << "RtApiJack::probeDeviceOpen: the requested sample rate (" << sampleRate << ") is different than the JACK server rate (" << jackRate << ").";
- errorText_ = errorStream_.str();
- return FAILURE;
- }
- stream_.sampleRate = jackRate;
-
- // Get the latency of the JACK port.
- ports = jack_get_ports( client, deviceName.c_str(), NULL, flag );
- if ( ports[ firstChannel ] ) {
- // Added by Ge Wang
- jack_latency_callback_mode_t cbmode = (mode == INPUT ? JackCaptureLatency : JackPlaybackLatency);
- // the range (usually the min and max are equal)
- jack_latency_range_t latrange; latrange.min = latrange.max = 0;
- // get the latency range
- jack_port_get_latency_range( jack_port_by_name( client, ports[firstChannel] ), cbmode, &latrange );
- // be optimistic, use the min!
- stream_.latency[mode] = latrange.min;
- //stream_.latency[mode] = jack_port_get_latency( jack_port_by_name( client, ports[ firstChannel ] ) );
- }
- free( ports );
-
- // The jack server always uses 32-bit floating-point data.
- stream_.deviceFormat[mode] = RTAUDIO_FLOAT32;
- stream_.userFormat = format;
-
- if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) stream_.userInterleaved = false;
- else stream_.userInterleaved = true;
-
- // Jack always uses non-interleaved buffers.
- stream_.deviceInterleaved[mode] = false;
-
- // Jack always provides host byte-ordered data.
- stream_.doByteSwap[mode] = false;
-
- // Get the buffer size. The buffer size and number of buffers
- // (periods) is set when the jack server is started.
- stream_.bufferSize = (int) jack_get_buffer_size( client );
- *bufferSize = stream_.bufferSize;
-
- stream_.nDeviceChannels[mode] = channels;
- stream_.nUserChannels[mode] = channels;
-
- // Set flags for buffer conversion.
- stream_.doConvertBuffer[mode] = false;
- if ( stream_.userFormat != stream_.deviceFormat[mode] )
- stream_.doConvertBuffer[mode] = true;
- if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] &&
- stream_.nUserChannels[mode] > 1 )
- stream_.doConvertBuffer[mode] = true;
-
- // Allocate our JackHandle structure for the stream.
- if ( handle == 0 ) {
- try {
- handle = new JackHandle;
- }
- catch ( std::bad_alloc& ) {
- errorText_ = "RtApiJack::probeDeviceOpen: error allocating JackHandle memory.";
- goto error;
- }
-
- if ( pthread_cond_init(&handle->condition, NULL) ) {
- errorText_ = "RtApiJack::probeDeviceOpen: error initializing pthread condition variable.";
- goto error;
- }
- stream_.apiHandle = (void *) handle;
- handle->client = client;
- }
- handle->deviceName[mode] = deviceName;
-
- // Allocate necessary internal buffers.
- unsigned long bufferBytes;
- bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat );
- stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 );
- if ( stream_.userBuffer[mode] == NULL ) {
- errorText_ = "RtApiJack::probeDeviceOpen: error allocating user buffer memory.";
- goto error;
- }
-
- if ( stream_.doConvertBuffer[mode] ) {
-
- bool makeBuffer = true;
- if ( mode == OUTPUT )
- bufferBytes = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] );
- else { // mode == INPUT
- bufferBytes = stream_.nDeviceChannels[1] * formatBytes( stream_.deviceFormat[1] );
- if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) {
- unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes(stream_.deviceFormat[0]);
- if ( bufferBytes < bytesOut ) makeBuffer = false;
- }
- }
-
- if ( makeBuffer ) {
- bufferBytes *= *bufferSize;
- if ( stream_.deviceBuffer ) free( stream_.deviceBuffer );
- stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 );
- if ( stream_.deviceBuffer == NULL ) {
- errorText_ = "RtApiJack::probeDeviceOpen: error allocating device buffer memory.";
- goto error;
- }
- }
- }
-
- // Allocate memory for the Jack ports (channels) identifiers.
- handle->ports[mode] = (jack_port_t **) malloc ( sizeof (jack_port_t *) * channels );
- if ( handle->ports[mode] == NULL ) {
- errorText_ = "RtApiJack::probeDeviceOpen: error allocating port memory.";
- goto error;
- }
-
- stream_.device[mode] = device;
- stream_.channelOffset[mode] = firstChannel;
- stream_.state = STREAM_STOPPED;
- stream_.callbackInfo.object = (void *) this;
-
- if ( stream_.mode == OUTPUT && mode == INPUT )
- // We had already set up the stream for output.
- stream_.mode = DUPLEX;
- else {
- stream_.mode = mode;
- jack_set_process_callback( handle->client, jackCallbackHandler, (void *) &stream_.callbackInfo );
- jack_set_xrun_callback( handle->client, jackXrun, (void *) &handle );
- jack_on_shutdown( handle->client, jackShutdown, (void *) &stream_.callbackInfo );
- }
-
- // Register our ports.
- char label[64];
- if ( mode == OUTPUT ) {
- for ( unsigned int i=0; i<stream_.nUserChannels[0]; i++ ) {
- snprintf( label, 64, "outport %d", i );
- handle->ports[0][i] = jack_port_register( handle->client, (const char *)label,
- JACK_DEFAULT_AUDIO_TYPE, JackPortIsOutput, 0 );
- }
- }
- else {
- for ( unsigned int i=0; i<stream_.nUserChannels[1]; i++ ) {
- snprintf( label, 64, "inport %d", i );
- handle->ports[1][i] = jack_port_register( handle->client, (const char *)label,
- JACK_DEFAULT_AUDIO_TYPE, JackPortIsInput, 0 );
- }
- }
-
- // Setup the buffer conversion information structure. We don't use
- // buffers to do channel offsets, so we override that parameter
- // here.
- if ( stream_.doConvertBuffer[mode] ) setConvertInfo( mode, 0 );
-
- return SUCCESS;
-
- error:
- if ( handle ) {
- pthread_cond_destroy( &handle->condition );
- jack_client_close( handle->client );
-
- if ( handle->ports[0] ) free( handle->ports[0] );
- if ( handle->ports[1] ) free( handle->ports[1] );
-
- delete handle;
- stream_.apiHandle = 0;
- }
-
- for ( int i=0; i<2; i++ ) {
- if ( stream_.userBuffer[i] ) {
- free( stream_.userBuffer[i] );
- stream_.userBuffer[i] = 0;
- }
- }
-
- if ( stream_.deviceBuffer ) {
- free( stream_.deviceBuffer );
- stream_.deviceBuffer = 0;
- }
-
- return FAILURE;
-}
-
-void RtApiJack :: closeStream( void )
-{
- if ( stream_.state == STREAM_CLOSED ) {
- errorText_ = "RtApiJack::closeStream(): no open stream to close!";
- error( RtAudioError::WARNING );
- return;
- }
-
- JackHandle *handle = (JackHandle *) stream_.apiHandle;
- if ( handle ) {
-
- if ( stream_.state == STREAM_RUNNING )
- jack_deactivate( handle->client );
-
- jack_client_close( handle->client );
- }
-
- if ( handle ) {
- if ( handle->ports[0] ) free( handle->ports[0] );
- if ( handle->ports[1] ) free( handle->ports[1] );
- pthread_cond_destroy( &handle->condition );
- delete handle;
- stream_.apiHandle = 0;
- }
-
- for ( int i=0; i<2; i++ ) {
- if ( stream_.userBuffer[i] ) {
- free( stream_.userBuffer[i] );
- stream_.userBuffer[i] = 0;
- }
- }
-
- if ( stream_.deviceBuffer ) {
- free( stream_.deviceBuffer );
- stream_.deviceBuffer = 0;
- }
-
- stream_.mode = UNINITIALIZED;
- stream_.state = STREAM_CLOSED;
-}
-
-void RtApiJack :: startStream( void )
-{
- verifyStream();
- if ( stream_.state == STREAM_RUNNING ) {
- errorText_ = "RtApiJack::startStream(): the stream is already running!";
- error( RtAudioError::WARNING );
- return;
- }
-
- JackHandle *handle = (JackHandle *) stream_.apiHandle;
- int result = jack_activate( handle->client );
- if ( result ) {
- errorText_ = "RtApiJack::startStream(): unable to activate JACK client!";
- goto unlock;
- }
-
- const char **ports;
-
- // Get the list of available ports.
- if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
- result = 1;
- ports = jack_get_ports( handle->client, handle->deviceName[0].c_str(), NULL, JackPortIsInput);
- if ( ports == NULL) {
- errorText_ = "RtApiJack::startStream(): error determining available JACK input ports!";
- goto unlock;
- }
-
- // Now make the port connections. Since RtAudio wasn't designed to
- // allow the user to select particular channels of a device, we'll
- // just open the first "nChannels" ports with offset.
- for ( unsigned int i=0; i<stream_.nUserChannels[0]; i++ ) {
- result = 1;
- if ( ports[ stream_.channelOffset[0] + i ] )
- result = jack_connect( handle->client, jack_port_name( handle->ports[0][i] ), ports[ stream_.channelOffset[0] + i ] );
- if ( result ) {
- free( ports );
- errorText_ = "RtApiJack::startStream(): error connecting output ports!";
- goto unlock;
- }
- }
- free(ports);
- }
-
- if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {
- result = 1;
- ports = jack_get_ports( handle->client, handle->deviceName[1].c_str(), NULL, JackPortIsOutput );
- if ( ports == NULL) {
- errorText_ = "RtApiJack::startStream(): error determining available JACK output ports!";
- goto unlock;
- }
-
- // Now make the port connections. See note above.
- for ( unsigned int i=0; i<stream_.nUserChannels[1]; i++ ) {
- result = 1;
- if ( ports[ stream_.channelOffset[1] + i ] )
- result = jack_connect( handle->client, ports[ stream_.channelOffset[1] + i ], jack_port_name( handle->ports[1][i] ) );
- if ( result ) {
- free( ports );
- errorText_ = "RtApiJack::startStream(): error connecting input ports!";
- goto unlock;
- }
- }
- free(ports);
- }
-
- handle->drainCounter = 0;
- handle->internalDrain = false;
- stream_.state = STREAM_RUNNING;
-
- unlock:
- if ( result == 0 ) return;
- error( RtAudioError::SYSTEM_ERROR );
-}
-
-void RtApiJack :: stopStream( void )
-{
- verifyStream();
- if ( stream_.state == STREAM_STOPPED ) {
- errorText_ = "RtApiJack::stopStream(): the stream is already stopped!";
- error( RtAudioError::WARNING );
- return;
- }
-
- JackHandle *handle = (JackHandle *) stream_.apiHandle;
- if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
-
- if ( handle->drainCounter == 0 ) {
- handle->drainCounter = 2;
- pthread_cond_wait( &handle->condition, &stream_.mutex ); // block until signaled
- }
- }
-
- jack_deactivate( handle->client );
- stream_.state = STREAM_STOPPED;
-}
-
-void RtApiJack :: abortStream( void )
-{
- verifyStream();
- if ( stream_.state == STREAM_STOPPED ) {
- errorText_ = "RtApiJack::abortStream(): the stream is already stopped!";
- error( RtAudioError::WARNING );
- return;
- }
-
- JackHandle *handle = (JackHandle *) stream_.apiHandle;
- handle->drainCounter = 2;
-
- stopStream();
-}
-
-// This function will be called by a spawned thread when the user
-// callback function signals that the stream should be stopped or
-// aborted. It is necessary to handle it this way because the
-// callbackEvent() function must return before the jack_deactivate()
-// function will return.
-static void *jackStopStream( void *ptr )
-{
- CallbackInfo *info = (CallbackInfo *) ptr;
- RtApiJack *object = (RtApiJack *) info->object;
-
- object->stopStream();
- pthread_exit( NULL );
-}
-
-bool RtApiJack :: callbackEvent( unsigned long nframes )
-{
- if ( stream_.state == STREAM_STOPPED || stream_.state == STREAM_STOPPING ) return SUCCESS;
- if ( stream_.state == STREAM_CLOSED ) {
- errorText_ = "RtApiCore::callbackEvent(): the stream is closed ... this shouldn't happen!";
- error( RtAudioError::WARNING );
- return FAILURE;
- }
- if ( stream_.bufferSize != nframes ) {
- errorText_ = "RtApiCore::callbackEvent(): the JACK buffer size has changed ... cannot process!";
- error( RtAudioError::WARNING );
- return FAILURE;
- }
-
- CallbackInfo *info = (CallbackInfo *) &stream_.callbackInfo;
- JackHandle *handle = (JackHandle *) stream_.apiHandle;
-
- // Check if we were draining the stream and signal is finished.
- if ( handle->drainCounter > 3 ) {
- ThreadHandle threadId;
-
- stream_.state = STREAM_STOPPING;
- if ( handle->internalDrain == true )
- pthread_create( &threadId, NULL, jackStopStream, info );
- else
- pthread_cond_signal( &handle->condition );
- return SUCCESS;
- }
-
- // Invoke user callback first, to get fresh output data.
- if ( handle->drainCounter == 0 ) {
- RtAudioCallback callback = (RtAudioCallback) info->callback;
- double streamTime = getStreamTime();
- RtAudioStreamStatus status = 0;
- if ( stream_.mode != INPUT && handle->xrun[0] == true ) {
- status |= RTAUDIO_OUTPUT_UNDERFLOW;
- handle->xrun[0] = false;
- }
- if ( stream_.mode != OUTPUT && handle->xrun[1] == true ) {
- status |= RTAUDIO_INPUT_OVERFLOW;
- handle->xrun[1] = false;
- }
- int cbReturnValue = callback( stream_.userBuffer[0], stream_.userBuffer[1],
- stream_.bufferSize, streamTime, status, info->userData );
- if ( cbReturnValue == 2 ) {
- stream_.state = STREAM_STOPPING;
- handle->drainCounter = 2;
- ThreadHandle id;
- pthread_create( &id, NULL, jackStopStream, info );
- return SUCCESS;
- }
- else if ( cbReturnValue == 1 ) {
- handle->drainCounter = 1;
- handle->internalDrain = true;
- }
- }
-
- jack_default_audio_sample_t *jackbuffer;
- unsigned long bufferBytes = nframes * sizeof( jack_default_audio_sample_t );
- if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
-
- if ( handle->drainCounter > 1 ) { // write zeros to the output stream
-
- for ( unsigned int i=0; i<stream_.nDeviceChannels[0]; i++ ) {
- jackbuffer = (jack_default_audio_sample_t *) jack_port_get_buffer( handle->ports[0][i], (jack_nframes_t) nframes );
- memset( jackbuffer, 0, bufferBytes );
- }
-
- }
- else if ( stream_.doConvertBuffer[0] ) {
-
- convertBuffer( stream_.deviceBuffer, stream_.userBuffer[0], stream_.convertInfo[0] );
-
- for ( unsigned int i=0; i<stream_.nDeviceChannels[0]; i++ ) {
- jackbuffer = (jack_default_audio_sample_t *) jack_port_get_buffer( handle->ports[0][i], (jack_nframes_t) nframes );
- memcpy( jackbuffer, &stream_.deviceBuffer[i*bufferBytes], bufferBytes );
- }
- }
- else { // no buffer conversion
- for ( unsigned int i=0; i<stream_.nUserChannels[0]; i++ ) {
- jackbuffer = (jack_default_audio_sample_t *) jack_port_get_buffer( handle->ports[0][i], (jack_nframes_t) nframes );
- memcpy( jackbuffer, &stream_.userBuffer[0][i*bufferBytes], bufferBytes );
- }
- }
- }
-
- // Don't bother draining input
- if ( handle->drainCounter ) {
- handle->drainCounter++;
- goto unlock;
- }
-
- if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {
-
- if ( stream_.doConvertBuffer[1] ) {
- for ( unsigned int i=0; i<stream_.nDeviceChannels[1]; i++ ) {
- jackbuffer = (jack_default_audio_sample_t *) jack_port_get_buffer( handle->ports[1][i], (jack_nframes_t) nframes );
- memcpy( &stream_.deviceBuffer[i*bufferBytes], jackbuffer, bufferBytes );
- }
- convertBuffer( stream_.userBuffer[1], stream_.deviceBuffer, stream_.convertInfo[1] );
- }
- else { // no buffer conversion
- for ( unsigned int i=0; i<stream_.nUserChannels[1]; i++ ) {
- jackbuffer = (jack_default_audio_sample_t *) jack_port_get_buffer( handle->ports[1][i], (jack_nframes_t) nframes );
- memcpy( &stream_.userBuffer[1][i*bufferBytes], jackbuffer, bufferBytes );
- }
- }
- }
-
- unlock:
- RtApi::tickStreamTime();
- return SUCCESS;
-}
- //******************** End of __UNIX_JACK__ *********************//
-#endif
-
-#if defined(__WINDOWS_ASIO__) // ASIO API on Windows
-
-// The ASIO API is designed around a callback scheme, so this
-// implementation is similar to that used for OS-X CoreAudio and Linux
-// Jack. The primary constraint with ASIO is that it only allows
-// access to a single driver at a time. Thus, it is not possible to
-// have more than one simultaneous RtAudio stream.
-//
-// This implementation also requires a number of external ASIO files
-// and a few global variables. The ASIO callback scheme does not
-// allow for the passing of user data, so we must create a global
-// pointer to our callbackInfo structure.
-//
-// On unix systems, we make use of a pthread condition variable.
-// Since there is no equivalent in Windows, I hacked something based
-// on information found in
-// http://www.cs.wustl.edu/~schmidt/win32-cv-1.html.
-
-#include "asiosys.h"
-#include "asio.h"
-#include "iasiothiscallresolver.h"
-#include "asiodrivers.h"
-#include <cmath>
-
-static AsioDrivers drivers;
-static ASIOCallbacks asioCallbacks;
-static ASIODriverInfo driverInfo;
-static CallbackInfo *asioCallbackInfo;
-static bool asioXRun;
-
-struct AsioHandle {
- int drainCounter; // Tracks callback counts when draining
- bool internalDrain; // Indicates if stop is initiated from callback or not.
- ASIOBufferInfo *bufferInfos;
- HANDLE condition;
-
- AsioHandle()
- :drainCounter(0), internalDrain(false), bufferInfos(0) {}
-};
-
-// Function declarations (definitions at end of section)
-static const char* getAsioErrorString( ASIOError result );
-static void sampleRateChanged( ASIOSampleRate sRate );
-static long asioMessages( long selector, long value, void* message, double* opt );
-
-RtApiAsio :: RtApiAsio()
-{
- // ASIO cannot run on a multi-threaded apartment. You can call
- // CoInitialize beforehand, but it must be for apartment 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 apartment. 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( &currentRate );
- 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>
-#include <math.h>
-
-//=============================================================================
-
-#define SAFE_RELEASE( objectPtr )\
-if ( objectPtr )\
-{\
- objectPtr->Release();\
- objectPtr = NULL;\
-}
-
-typedef HANDLE ( __stdcall *TAvSetMmThreadCharacteristicsPtr )( LPCWSTR TaskName, LPDWORD TaskIndex );
-
-//-----------------------------------------------------------------------------
-
-// WASAPI dictates stream sample rate, format, channel count, and in some cases, buffer size.
-// Therefore we must perform all necessary conversions to user buffers in order to satisfy these
-// requirements. WasapiBuffer ring buffers are used between HwIn->UserIn and UserOut->HwOut to
-// provide intermediate storage for read / write synchronization.
-class WasapiBuffer
-{
-public:
- WasapiBuffer()
- : buffer_( NULL ),
- bufferSize_( 0 ),
- inIndex_( 0 ),
- outIndex_( 0 ) {}
-
- ~WasapiBuffer() {
- free( buffer_ );
- }
-
- // sets the length of the internal ring buffer
- void setBufferSize( unsigned int bufferSize, unsigned int formatBytes ) {
- free( buffer_ );
-
- buffer_ = ( char* ) calloc( bufferSize, formatBytes );
-
- bufferSize_ = bufferSize;
- inIndex_ = 0;
- outIndex_ = 0;
- }
-
- // attempt to push a buffer into the ring buffer at the current "in" index
- bool pushBuffer( char* buffer, unsigned int bufferSize, RtAudioFormat format )
- {
- if ( !buffer || // incoming buffer is NULL
- bufferSize == 0 || // incoming buffer has no data
- bufferSize > bufferSize_ ) // incoming buffer too large
- {
- return false;
- }
-
- unsigned int relOutIndex = outIndex_;
- unsigned int inIndexEnd = inIndex_ + bufferSize;
- if ( relOutIndex < inIndex_ && inIndexEnd >= bufferSize_ ) {
- relOutIndex += bufferSize_;
- }
-
- // "in" index can end on the "out" index but cannot begin at it
- if ( inIndex_ <= relOutIndex && inIndexEnd > relOutIndex ) {
- return false; // not enough space between "in" index and "out" index
- }
-
- // copy buffer from external to internal
- int fromZeroSize = inIndex_ + bufferSize - bufferSize_;
- fromZeroSize = fromZeroSize < 0 ? 0 : fromZeroSize;
- int fromInSize = bufferSize - fromZeroSize;
-
- switch( format )
- {
- case RTAUDIO_SINT8:
- memcpy( &( ( char* ) buffer_ )[inIndex_], buffer, fromInSize * sizeof( char ) );
- memcpy( buffer_, &( ( char* ) buffer )[fromInSize], fromZeroSize * sizeof( char ) );
- break;
- case RTAUDIO_SINT16:
- memcpy( &( ( short* ) buffer_ )[inIndex_], buffer, fromInSize * sizeof( short ) );
- memcpy( buffer_, &( ( short* ) buffer )[fromInSize], fromZeroSize * sizeof( short ) );
- break;
- case RTAUDIO_SINT24:
- memcpy( &( ( S24* ) buffer_ )[inIndex_], buffer, fromInSize * sizeof( S24 ) );
- memcpy( buffer_, &( ( S24* ) buffer )[fromInSize], fromZeroSize * sizeof( S24 ) );
- break;
- case RTAUDIO_SINT32:
- memcpy( &( ( int* ) buffer_ )[inIndex_], buffer, fromInSize * sizeof( int ) );
- memcpy( buffer_, &( ( int* ) buffer )[fromInSize], fromZeroSize * sizeof( int ) );
- break;
- case RTAUDIO_FLOAT32:
- memcpy( &( ( float* ) buffer_ )[inIndex_], buffer, fromInSize * sizeof( float ) );
- memcpy( buffer_, &( ( float* ) buffer )[fromInSize], fromZeroSize * sizeof( float ) );
- break;
- case RTAUDIO_FLOAT64:
- memcpy( &( ( double* ) buffer_ )[inIndex_], buffer, fromInSize * sizeof( double ) );
- memcpy( buffer_, &( ( double* ) buffer )[fromInSize], fromZeroSize * sizeof( double ) );
- break;
- }
-
- // update "in" index
- inIndex_ += bufferSize;
- inIndex_ %= bufferSize_;
-
- return true;
- }
-
- // attempt to pull a buffer from the ring buffer from the current "out" index
- bool pullBuffer( char* buffer, unsigned int bufferSize, RtAudioFormat format )
- {
- if ( !buffer || // incoming buffer is NULL
- bufferSize == 0 || // incoming buffer has no data
- bufferSize > bufferSize_ ) // incoming buffer too large
- {
- return false;
- }
-
- unsigned int relInIndex = inIndex_;
- unsigned int outIndexEnd = outIndex_ + bufferSize;
- if ( relInIndex < outIndex_ && outIndexEnd >= bufferSize_ ) {
- relInIndex += bufferSize_;
- }
-
- // "out" index can begin at and end on the "in" index
- if ( outIndex_ < relInIndex && outIndexEnd > relInIndex ) {
- return false; // not enough space between "out" index and "in" index
- }
-
- // copy buffer from internal to external
- int fromZeroSize = outIndex_ + bufferSize - bufferSize_;
- fromZeroSize = fromZeroSize < 0 ? 0 : fromZeroSize;
- int fromOutSize = bufferSize - fromZeroSize;
-
- switch( format )
- {
- case RTAUDIO_SINT8:
- memcpy( buffer, &( ( char* ) buffer_ )[outIndex_], fromOutSize * sizeof( char ) );
- memcpy( &( ( char* ) buffer )[fromOutSize], buffer_, fromZeroSize * sizeof( char ) );
- break;
- case RTAUDIO_SINT16:
- memcpy( buffer, &( ( short* ) buffer_ )[outIndex_], fromOutSize * sizeof( short ) );
- memcpy( &( ( short* ) buffer )[fromOutSize], buffer_, fromZeroSize * sizeof( short ) );
- break;
- case RTAUDIO_SINT24:
- memcpy( buffer, &( ( S24* ) buffer_ )[outIndex_], fromOutSize * sizeof( S24 ) );
- memcpy( &( ( S24* ) buffer )[fromOutSize], buffer_, fromZeroSize * sizeof( S24 ) );
- break;
- case RTAUDIO_SINT32:
- memcpy( buffer, &( ( int* ) buffer_ )[outIndex_], fromOutSize * sizeof( int ) );
- memcpy( &( ( int* ) buffer )[fromOutSize], buffer_, fromZeroSize * sizeof( int ) );
- break;
- case RTAUDIO_FLOAT32:
- memcpy( buffer, &( ( float* ) buffer_ )[outIndex_], fromOutSize * sizeof( float ) );
- memcpy( &( ( float* ) buffer )[fromOutSize], buffer_, fromZeroSize * sizeof( float ) );
- break;
- case RTAUDIO_FLOAT64:
- memcpy( buffer, &( ( double* ) buffer_ )[outIndex_], fromOutSize * sizeof( double ) );
- memcpy( &( ( double* ) buffer )[fromOutSize], buffer_, fromZeroSize * sizeof( double ) );
- break;
- }
-
- // update "out" index
- outIndex_ += bufferSize;
- outIndex_ %= bufferSize_;
-
- return true;
- }
-
-private:
- char* buffer_;
- unsigned int bufferSize_;
- unsigned int inIndex_;
- unsigned int outIndex_;
-};
-
-//-----------------------------------------------------------------------------
-
-// In order to satisfy WASAPI's buffer requirements, we need a means of converting sample rate
-// between HW and the user. The convertBufferWasapi function is used to perform this conversion
-// between HwIn->UserIn and UserOut->HwOut during the stream callback loop.
-// This sample rate converter favors speed over quality, and works best with conversions between
-// one rate and its multiple.
-void convertBufferWasapi( char* outBuffer,
- const char* inBuffer,
- const unsigned int& channelCount,
- const unsigned int& inSampleRate,
- const unsigned int& outSampleRate,
- const unsigned int& inSampleCount,
- unsigned int& outSampleCount,
- const RtAudioFormat& format )
-{
- // calculate the new outSampleCount and relative sampleStep
- float sampleRatio = ( float ) outSampleRate / inSampleRate;
- float sampleStep = 1.0f / sampleRatio;
- float inSampleFraction = 0.0f;
-
- outSampleCount = ( unsigned int ) roundf( inSampleCount * sampleRatio );
-
- // frame-by-frame, copy each relative input sample into it's corresponding output sample
- for ( unsigned int outSample = 0; outSample < outSampleCount; outSample++ )
- {
- unsigned int inSample = ( unsigned int ) inSampleFraction;
-
- switch ( format )
- {
- case RTAUDIO_SINT8:
- memcpy( &( ( char* ) outBuffer )[ outSample * channelCount ], &( ( char* ) inBuffer )[ inSample * channelCount ], channelCount * sizeof( char ) );
- break;
- case RTAUDIO_SINT16:
- memcpy( &( ( short* ) outBuffer )[ outSample * channelCount ], &( ( short* ) inBuffer )[ inSample * channelCount ], channelCount * sizeof( short ) );
- break;
- case RTAUDIO_SINT24:
- memcpy( &( ( S24* ) outBuffer )[ outSample * channelCount ], &( ( S24* ) inBuffer )[ inSample * channelCount ], channelCount * sizeof( S24 ) );
- break;
- case RTAUDIO_SINT32:
- memcpy( &( ( int* ) outBuffer )[ outSample * channelCount ], &( ( int* ) inBuffer )[ inSample * channelCount ], channelCount * sizeof( int ) );
- break;
- case RTAUDIO_FLOAT32:
- memcpy( &( ( float* ) outBuffer )[ outSample * channelCount ], &( ( float* ) inBuffer )[ inSample * channelCount ], channelCount * sizeof( float ) );
- break;
- case RTAUDIO_FLOAT64:
- memcpy( &( ( double* ) outBuffer )[ outSample * channelCount ], &( ( double* ) inBuffer )[ inSample * channelCount ], channelCount * sizeof( double ) );
- break;
- }
-
- // jump to next in sample
- inSampleFraction += sampleStep;
- }
-}
-
-//-----------------------------------------------------------------------------
-
-// A structure to hold various information related to the WASAPI implementation.
-struct WasapiHandle
-{
- IAudioClient* captureAudioClient;
- IAudioClient* renderAudioClient;
- IAudioCaptureClient* captureClient;
- IAudioRenderClient* renderClient;
- HANDLE captureEvent;
- HANDLE renderEvent;
-
- WasapiHandle()
- : captureAudioClient( NULL ),
- renderAudioClient( NULL ),
- captureClient( NULL ),
- renderClient( NULL ),
- captureEvent( NULL ),
- renderEvent( NULL ) {}
-};
-
-//=============================================================================
-
-RtApiWasapi::RtApiWasapi()
- : coInitialized_( false ), deviceEnumerator_( NULL )
-{
- // WASAPI can run either apartment or multi-threaded
- HRESULT hr = CoInitialize( NULL );
- if ( !FAILED( hr ) )
- coInitialized_ = true;
-
- // Instantiate device enumerator
- hr = CoCreateInstance( __uuidof( MMDeviceEnumerator ), NULL,
- CLSCTX_ALL, __uuidof( IMMDeviceEnumerator ),
- ( void** ) &deviceEnumerator_ );
-
- if ( FAILED( hr ) ) {
- errorText_ = "RtApiWasapi::RtApiWasapi: Unable to instantiate device enumerator";
- error( RtAudioError::DRIVER_ERROR );
- }
-}
-
-//-----------------------------------------------------------------------------
-
-RtApiWasapi::~RtApiWasapi()
-{
- if ( stream_.state != STREAM_CLOSED )
- closeStream();
-
- SAFE_RELEASE( deviceEnumerator_ );
-
- // If this object previously called CoInitialize()
- if ( coInitialized_ )
- CoUninitialize();
-}
-
-//=============================================================================
-
-unsigned int RtApiWasapi::getDeviceCount( void )
-{
- unsigned int captureDeviceCount = 0;
- unsigned int renderDeviceCount = 0;
-
- IMMDeviceCollection* captureDevices = NULL;
- IMMDeviceCollection* renderDevices = NULL;
-
- // Count capture devices
- errorText_.clear();
- HRESULT hr = deviceEnumerator_->EnumAudioEndpoints( eCapture, DEVICE_STATE_ACTIVE, &captureDevices );
- if ( FAILED( hr ) ) {
- errorText_ = "RtApiWasapi::getDeviceCount: Unable to retrieve capture device collection.";
- goto Exit;
- }
-
- hr = captureDevices->GetCount( &captureDeviceCount );
- if ( FAILED( hr ) ) {
- errorText_ = "RtApiWasapi::getDeviceCount: Unable to retrieve capture device count.";
- goto Exit;
- }
-
- // Count render devices
- hr = deviceEnumerator_->EnumAudioEndpoints( eRender, DEVICE_STATE_ACTIVE, &renderDevices );
- if ( FAILED( hr ) ) {
- errorText_ = "RtApiWasapi::getDeviceCount: Unable to retrieve render device collection.";
- goto Exit;
- }
-
- hr = renderDevices->GetCount( &renderDeviceCount );
- if ( FAILED( hr ) ) {
- errorText_ = "RtApiWasapi::getDeviceCount: Unable to retrieve render device count.";
- goto Exit;
- }
-
-Exit:
- // release all references
- SAFE_RELEASE( captureDevices );
- SAFE_RELEASE( renderDevices );
-
- if ( errorText_.empty() )
- return captureDeviceCount + renderDeviceCount;
-
- error( RtAudioError::DRIVER_ERROR );
- return 0;
-}
-
-//-----------------------------------------------------------------------------
-
-RtAudio::DeviceInfo RtApiWasapi::getDeviceInfo( unsigned int device )
-{
- RtAudio::DeviceInfo info;
- unsigned int captureDeviceCount = 0;
- unsigned int renderDeviceCount = 0;
- std::string defaultDeviceName;
- bool isCaptureDevice = false;
-
- PROPVARIANT deviceNameProp;
- PROPVARIANT defaultDeviceNameProp;
-
- IMMDeviceCollection* captureDevices = NULL;
- IMMDeviceCollection* renderDevices = NULL;
- IMMDevice* devicePtr = NULL;
- IMMDevice* defaultDevicePtr = NULL;
- IAudioClient* audioClient = NULL;
- IPropertyStore* devicePropStore = NULL;
- IPropertyStore* defaultDevicePropStore = NULL;
-
- WAVEFORMATEX* deviceFormat = NULL;
- WAVEFORMATEX* closestMatchFormat = NULL;
-
- // probed
- info.probed = false;
-
- // Count capture devices
- errorText_.clear();
- RtAudioError::Type errorType = RtAudioError::DRIVER_ERROR;
- HRESULT hr = deviceEnumerator_->EnumAudioEndpoints( eCapture, DEVICE_STATE_ACTIVE, &captureDevices );
- if ( FAILED( hr ) ) {
- errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve capture device collection.";
- goto Exit;
- }
-
- hr = captureDevices->GetCount( &captureDeviceCount );
- if ( FAILED( hr ) ) {
- errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve capture device count.";
- goto Exit;
- }
-
- // Count render devices
- hr = deviceEnumerator_->EnumAudioEndpoints( eRender, DEVICE_STATE_ACTIVE, &renderDevices );
- if ( FAILED( hr ) ) {
- errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve render device collection.";
- goto Exit;
- }
-
- hr = renderDevices->GetCount( &renderDeviceCount );
- if ( FAILED( hr ) ) {
- errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve render device count.";
- goto Exit;
- }
-
- // validate device index
- if ( device >= captureDeviceCount + renderDeviceCount ) {
- errorText_ = "RtApiWasapi::getDeviceInfo: Invalid device index.";
- errorType = RtAudioError::INVALID_USE;
- goto Exit;
- }
-
- // determine whether index falls within capture or render devices
- if ( device >= renderDeviceCount ) {
- hr = captureDevices->Item( device - renderDeviceCount, &devicePtr );
- if ( FAILED( hr ) ) {
- errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve capture device handle.";
- goto Exit;
- }
- isCaptureDevice = true;
- }
- else {
- hr = renderDevices->Item( device, &devicePtr );
- if ( FAILED( hr ) ) {
- errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve render device handle.";
- goto Exit;
- }
- isCaptureDevice = false;
- }
-
- // get default device name
- if ( isCaptureDevice ) {
- hr = deviceEnumerator_->GetDefaultAudioEndpoint( eCapture, eConsole, &defaultDevicePtr );
- if ( FAILED( hr ) ) {
- errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve default capture device handle.";
- goto Exit;
- }
- }
- else {
- hr = deviceEnumerator_->GetDefaultAudioEndpoint( eRender, eConsole, &defaultDevicePtr );
- if ( FAILED( hr ) ) {
- errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve default render device handle.";
- goto Exit;
- }
- }
-
- hr = defaultDevicePtr->OpenPropertyStore( STGM_READ, &defaultDevicePropStore );
- if ( FAILED( hr ) ) {
- errorText_ = "RtApiWasapi::getDeviceInfo: Unable to open default device property store.";
- goto Exit;
- }
- PropVariantInit( &defaultDeviceNameProp );
-
- hr = defaultDevicePropStore->GetValue( PKEY_Device_FriendlyName, &defaultDeviceNameProp );
- if ( FAILED( hr ) ) {
- errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve default device property: PKEY_Device_FriendlyName.";
- goto Exit;
- }
-
- defaultDeviceName = convertCharPointerToStdString(defaultDeviceNameProp.pwszVal);
-
- // name
- hr = devicePtr->OpenPropertyStore( STGM_READ, &devicePropStore );
- if ( FAILED( hr ) ) {
- errorText_ = "RtApiWasapi::getDeviceInfo: Unable to open device property store.";
- goto Exit;
- }
-
- PropVariantInit( &deviceNameProp );
-
- hr = devicePropStore->GetValue( PKEY_Device_FriendlyName, &deviceNameProp );
- if ( FAILED( hr ) ) {
- errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve device property: PKEY_Device_FriendlyName.";
- goto Exit;
- }
-
- info.name =convertCharPointerToStdString(deviceNameProp.pwszVal);
-
- // is default
- if ( isCaptureDevice ) {
- info.isDefaultInput = info.name == defaultDeviceName;
- info.isDefaultOutput = false;
- }
- else {
- info.isDefaultInput = false;
- info.isDefaultOutput = info.name == defaultDeviceName;
- }
-
- // channel count
- hr = devicePtr->Activate( __uuidof( IAudioClient ), CLSCTX_ALL, NULL, ( void** ) &audioClient );
- if ( FAILED( hr ) ) {
- errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve device audio client.";
- goto Exit;
- }
-
- hr = audioClient->GetMixFormat( &deviceFormat );
- if ( FAILED( hr ) ) {
- errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve device mix format.";
- goto Exit;
- }
-
- if ( isCaptureDevice ) {
- info.inputChannels = deviceFormat->nChannels;
- info.outputChannels = 0;
- info.duplexChannels = 0;
- }
- else {
- info.inputChannels = 0;
- info.outputChannels = deviceFormat->nChannels;
- info.duplexChannels = 0;
- }
-
- // sample rates
- info.sampleRates.clear();
-
- // allow support for all sample rates as we have a built-in sample rate converter
- for ( unsigned int i = 0; i < MAX_SAMPLE_RATES; i++ ) {
- info.sampleRates.push_back( SAMPLE_RATES[i] );
- }
- info.preferredSampleRate = deviceFormat->nSamplesPerSec;
-
- // native format
- info.nativeFormats = 0;
-
- if ( deviceFormat->wFormatTag == WAVE_FORMAT_IEEE_FLOAT ||
- ( deviceFormat->wFormatTag == WAVE_FORMAT_EXTENSIBLE &&
- ( ( WAVEFORMATEXTENSIBLE* ) deviceFormat )->SubFormat == KSDATAFORMAT_SUBTYPE_IEEE_FLOAT ) )
- {
- if ( deviceFormat->wBitsPerSample == 32 ) {
- info.nativeFormats |= RTAUDIO_FLOAT32;
- }
- else if ( deviceFormat->wBitsPerSample == 64 ) {
- info.nativeFormats |= RTAUDIO_FLOAT64;
- }
- }
- else if ( deviceFormat->wFormatTag == WAVE_FORMAT_PCM ||
- ( deviceFormat->wFormatTag == WAVE_FORMAT_EXTENSIBLE &&
- ( ( WAVEFORMATEXTENSIBLE* ) deviceFormat )->SubFormat == KSDATAFORMAT_SUBTYPE_PCM ) )
- {
- if ( deviceFormat->wBitsPerSample == 8 ) {
- info.nativeFormats |= RTAUDIO_SINT8;
- }
- else if ( deviceFormat->wBitsPerSample == 16 ) {
- info.nativeFormats |= RTAUDIO_SINT16;
- }
- else if ( deviceFormat->wBitsPerSample == 24 ) {
- info.nativeFormats |= RTAUDIO_SINT24;
- }
- else if ( deviceFormat->wBitsPerSample == 32 ) {
- info.nativeFormats |= RTAUDIO_SINT32;
- }
- }
-
- // probed
- info.probed = true;
-
-Exit:
- // release all references
- PropVariantClear( &deviceNameProp );
- PropVariantClear( &defaultDeviceNameProp );
-
- SAFE_RELEASE( captureDevices );
- SAFE_RELEASE( renderDevices );
- SAFE_RELEASE( devicePtr );
- SAFE_RELEASE( defaultDevicePtr );
- SAFE_RELEASE( audioClient );
- SAFE_RELEASE( devicePropStore );
- SAFE_RELEASE( defaultDevicePropStore );
-
- CoTaskMemFree( deviceFormat );
- CoTaskMemFree( closestMatchFormat );
-
- if ( !errorText_.empty() )
- error( errorType );
- return info;
-}
-
-//-----------------------------------------------------------------------------
-
-unsigned int RtApiWasapi::getDefaultOutputDevice( void )
-{
- for ( unsigned int i = 0; i < getDeviceCount(); i++ ) {
- if ( getDeviceInfo( i ).isDefaultOutput ) {
- return i;
- }
- }
-
- return 0;
-}
-
-//-----------------------------------------------------------------------------
-
-unsigned int RtApiWasapi::getDefaultInputDevice( void )
-{
- for ( unsigned int i = 0; i < getDeviceCount(); i++ ) {
- if ( getDeviceInfo( i ).isDefaultInput ) {
- return i;
- }
- }
-
- return 0;
-}
-
-//-----------------------------------------------------------------------------
-
-void RtApiWasapi::closeStream( void )
-{
- if ( stream_.state == STREAM_CLOSED ) {
- errorText_ = "RtApiWasapi::closeStream: No open stream to close.";
- error( RtAudioError::WARNING );
- return;
- }
-
- if ( stream_.state != STREAM_STOPPED )
- stopStream();
-
- // clean up stream memory
- SAFE_RELEASE( ( ( WasapiHandle* ) stream_.apiHandle )->captureAudioClient )
- SAFE_RELEASE( ( ( WasapiHandle* ) stream_.apiHandle )->renderAudioClient )
-
- SAFE_RELEASE( ( ( WasapiHandle* ) stream_.apiHandle )->captureClient )
- SAFE_RELEASE( ( ( WasapiHandle* ) stream_.apiHandle )->renderClient )
-
- if ( ( ( WasapiHandle* ) stream_.apiHandle )->captureEvent )
- CloseHandle( ( ( WasapiHandle* ) stream_.apiHandle )->captureEvent );
-
- if ( ( ( WasapiHandle* ) stream_.apiHandle )->renderEvent )
- CloseHandle( ( ( WasapiHandle* ) stream_.apiHandle )->renderEvent );
-
- delete ( WasapiHandle* ) stream_.apiHandle;
- stream_.apiHandle = NULL;
-
- for ( int i = 0; i < 2; i++ ) {
- if ( stream_.userBuffer[i] ) {
- free( stream_.userBuffer[i] );
- stream_.userBuffer[i] = 0;
- }
- }
-
- if ( stream_.deviceBuffer ) {
- free( stream_.deviceBuffer );
- stream_.deviceBuffer = 0;
- }
-
- // update stream state
- stream_.state = STREAM_CLOSED;
-}
-
-//-----------------------------------------------------------------------------
-
-void RtApiWasapi::startStream( void )
-{
- verifyStream();
-
- if ( stream_.state == STREAM_RUNNING ) {
- errorText_ = "RtApiWasapi::startStream: The stream is already running.";
- error( RtAudioError::WARNING );
- return;
- }
-
- // update stream state
- stream_.state = STREAM_RUNNING;
-
- // create WASAPI stream thread
- stream_.callbackInfo.thread = ( ThreadHandle ) CreateThread( NULL, 0, runWasapiThread, this, CREATE_SUSPENDED, NULL );
-
- if ( !stream_.callbackInfo.thread ) {
- errorText_ = "RtApiWasapi::startStream: Unable to instantiate callback thread.";
- error( RtAudioError::THREAD_ERROR );
- }
- else {
- SetThreadPriority( ( void* ) stream_.callbackInfo.thread, stream_.callbackInfo.priority );
- ResumeThread( ( void* ) stream_.callbackInfo.thread );
- }
-}
-
-//-----------------------------------------------------------------------------
-
-void RtApiWasapi::stopStream( void )
-{
- verifyStream();
-
- if ( stream_.state == STREAM_STOPPED ) {
- errorText_ = "RtApiWasapi::stopStream: The stream is already stopped.";
- error( RtAudioError::WARNING );
- return;
- }
-
- // inform stream thread by setting stream state to STREAM_STOPPING
- stream_.state = STREAM_STOPPING;
-
- // wait until stream thread is stopped
- while( stream_.state != STREAM_STOPPED ) {
- Sleep( 1 );
- }
-
- // Wait for the last buffer to play before stopping.
- Sleep( 1000 * stream_.bufferSize / stream_.sampleRate );
-
- // stop capture client if applicable
- if ( ( ( WasapiHandle* ) stream_.apiHandle )->captureAudioClient ) {
- HRESULT hr = ( ( WasapiHandle* ) stream_.apiHandle )->captureAudioClient->Stop();
- if ( FAILED( hr ) ) {
- errorText_ = "RtApiWasapi::stopStream: Unable to stop capture stream.";
- error( RtAudioError::DRIVER_ERROR );
- return;
- }
- }
-
- // stop render client if applicable
- if ( ( ( WasapiHandle* ) stream_.apiHandle )->renderAudioClient ) {
- HRESULT hr = ( ( WasapiHandle* ) stream_.apiHandle )->renderAudioClient->Stop();
- if ( FAILED( hr ) ) {
- errorText_ = "RtApiWasapi::stopStream: Unable to stop render stream.";
- error( RtAudioError::DRIVER_ERROR );
- return;
- }
- }
-
- // close thread handle
- if ( stream_.callbackInfo.thread && !CloseHandle( ( void* ) stream_.callbackInfo.thread ) ) {
- errorText_ = "RtApiWasapi::stopStream: Unable to close callback thread.";
- error( RtAudioError::THREAD_ERROR );
- return;
- }
-
- stream_.callbackInfo.thread = (ThreadHandle) NULL;
-}
-
-//-----------------------------------------------------------------------------
-
-void RtApiWasapi::abortStream( void )
-{
- verifyStream();
-
- if ( stream_.state == STREAM_STOPPED ) {
- errorText_ = "RtApiWasapi::abortStream: The stream is already stopped.";
- error( RtAudioError::WARNING );
- return;
- }
-
- // inform stream thread by setting stream state to STREAM_STOPPING
- stream_.state = STREAM_STOPPING;
-
- // wait until stream thread is stopped
- while ( stream_.state != STREAM_STOPPED ) {
- Sleep( 1 );
- }
-
- // stop capture client if applicable
- if ( ( ( WasapiHandle* ) stream_.apiHandle )->captureAudioClient ) {
- HRESULT hr = ( ( WasapiHandle* ) stream_.apiHandle )->captureAudioClient->Stop();
- if ( FAILED( hr ) ) {
- errorText_ = "RtApiWasapi::abortStream: Unable to stop capture stream.";
- error( RtAudioError::DRIVER_ERROR );
- return;
- }
- }
-
- // stop render client if applicable
- if ( ( ( WasapiHandle* ) stream_.apiHandle )->renderAudioClient ) {
- HRESULT hr = ( ( WasapiHandle* ) stream_.apiHandle )->renderAudioClient->Stop();
- if ( FAILED( hr ) ) {
- errorText_ = "RtApiWasapi::abortStream: Unable to stop render stream.";
- error( RtAudioError::DRIVER_ERROR );
- return;
- }
- }
-
- // close thread handle
- if ( stream_.callbackInfo.thread && !CloseHandle( ( void* ) stream_.callbackInfo.thread ) ) {
- errorText_ = "RtApiWasapi::abortStream: Unable to close callback thread.";
- error( RtAudioError::THREAD_ERROR );
- return;
- }
-
- stream_.callbackInfo.thread = (ThreadHandle) NULL;
-}
-
-//-----------------------------------------------------------------------------
-
-bool RtApiWasapi::probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,
- unsigned int firstChannel, unsigned int sampleRate,
- RtAudioFormat format, unsigned int* bufferSize,
- RtAudio::StreamOptions* options )
-{
- bool methodResult = FAILURE;
- unsigned int captureDeviceCount = 0;
- unsigned int renderDeviceCount = 0;
-
- IMMDeviceCollection* captureDevices = NULL;
- IMMDeviceCollection* renderDevices = NULL;
- IMMDevice* devicePtr = NULL;
- WAVEFORMATEX* deviceFormat = NULL;
- unsigned int bufferBytes;
- stream_.state = STREAM_STOPPED;
-
- // create API Handle if not already created
- if ( !stream_.apiHandle )
- stream_.apiHandle = ( void* ) new WasapiHandle();
-
- // Count capture devices
- errorText_.clear();
- RtAudioError::Type errorType = RtAudioError::DRIVER_ERROR;
- HRESULT hr = deviceEnumerator_->EnumAudioEndpoints( eCapture, DEVICE_STATE_ACTIVE, &captureDevices );
- if ( FAILED( hr ) ) {
- errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve capture device collection.";
- goto Exit;
- }
-
- hr = captureDevices->GetCount( &captureDeviceCount );
- if ( FAILED( hr ) ) {
- errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve capture device count.";
- goto Exit;
- }
-
- // Count render devices
- hr = deviceEnumerator_->EnumAudioEndpoints( eRender, DEVICE_STATE_ACTIVE, &renderDevices );
- if ( FAILED( hr ) ) {
- errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve render device collection.";
- goto Exit;
- }
-
- hr = renderDevices->GetCount( &renderDeviceCount );
- if ( FAILED( hr ) ) {
- errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve render device count.";
- goto Exit;
- }
-
- // validate device index
- if ( device >= captureDeviceCount + renderDeviceCount ) {
- errorType = RtAudioError::INVALID_USE;
- errorText_ = "RtApiWasapi::probeDeviceOpen: Invalid device index.";
- goto Exit;
- }
-
- // determine whether index falls within capture or render devices
- if ( device >= renderDeviceCount ) {
- if ( mode != INPUT ) {
- errorType = RtAudioError::INVALID_USE;
- errorText_ = "RtApiWasapi::probeDeviceOpen: Capture device selected as output device.";
- goto Exit;
- }
-
- // retrieve captureAudioClient from devicePtr
- IAudioClient*& captureAudioClient = ( ( WasapiHandle* ) stream_.apiHandle )->captureAudioClient;
-
- hr = captureDevices->Item( device - renderDeviceCount, &devicePtr );
- if ( FAILED( hr ) ) {
- errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve capture device handle.";
- goto Exit;
- }
-
- hr = devicePtr->Activate( __uuidof( IAudioClient ), CLSCTX_ALL,
- NULL, ( void** ) &captureAudioClient );
- if ( FAILED( hr ) ) {
- errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve device audio client.";
- goto Exit;
- }
-
- hr = captureAudioClient->GetMixFormat( &deviceFormat );
- if ( FAILED( hr ) ) {
- errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve device mix format.";
- goto Exit;
- }
-
- stream_.nDeviceChannels[mode] = deviceFormat->nChannels;
- captureAudioClient->GetStreamLatency( ( long long* ) &stream_.latency[mode] );
- }
- else {
- if ( mode != OUTPUT ) {
- errorType = RtAudioError::INVALID_USE;
- errorText_ = "RtApiWasapi::probeDeviceOpen: Render device selected as input device.";
- goto Exit;
- }
-
- // retrieve renderAudioClient from devicePtr
- IAudioClient*& renderAudioClient = ( ( WasapiHandle* ) stream_.apiHandle )->renderAudioClient;
-
- hr = renderDevices->Item( device, &devicePtr );
- if ( FAILED( hr ) ) {
- errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve render device handle.";
- goto Exit;
- }
-
- hr = devicePtr->Activate( __uuidof( IAudioClient ), CLSCTX_ALL,
- NULL, ( void** ) &renderAudioClient );
- if ( FAILED( hr ) ) {
- errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve device audio client.";
- goto Exit;
- }
-
- hr = renderAudioClient->GetMixFormat( &deviceFormat );
- if ( FAILED( hr ) ) {
- errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve device mix format.";
- goto Exit;
- }
-
- stream_.nDeviceChannels[mode] = deviceFormat->nChannels;
- renderAudioClient->GetStreamLatency( ( long long* ) &stream_.latency[mode] );
- }
-
- // fill stream data
- if ( ( stream_.mode == OUTPUT && mode == INPUT ) ||
- ( stream_.mode == INPUT && mode == OUTPUT ) ) {
- stream_.mode = DUPLEX;
- }
- else {
- stream_.mode = mode;
- }
-
- stream_.device[mode] = device;
- stream_.doByteSwap[mode] = false;
- stream_.sampleRate = sampleRate;
- stream_.bufferSize = *bufferSize;
- stream_.nBuffers = 1;
- stream_.nUserChannels[mode] = channels;
- stream_.channelOffset[mode] = firstChannel;
- stream_.userFormat = format;
- stream_.deviceFormat[mode] = getDeviceInfo( device ).nativeFormats;
-
- if ( options && options->flags & RTAUDIO_NONINTERLEAVED )
- stream_.userInterleaved = false;
- else
- stream_.userInterleaved = true;
- stream_.deviceInterleaved[mode] = true;
-
- // Set flags for buffer conversion.
- stream_.doConvertBuffer[mode] = false;
- if ( stream_.userFormat != stream_.deviceFormat[mode] ||
- stream_.nUserChannels != stream_.nDeviceChannels )
- stream_.doConvertBuffer[mode] = true;
- else if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] &&
- stream_.nUserChannels[mode] > 1 )
- stream_.doConvertBuffer[mode] = true;
-
- if ( stream_.doConvertBuffer[mode] )
- setConvertInfo( mode, 0 );
-
- // Allocate necessary internal buffers
- bufferBytes = stream_.nUserChannels[mode] * stream_.bufferSize * formatBytes( stream_.userFormat );
-
- stream_.userBuffer[mode] = ( char* ) calloc( bufferBytes, 1 );
- if ( !stream_.userBuffer[mode] ) {
- errorType = RtAudioError::MEMORY_ERROR;
- errorText_ = "RtApiWasapi::probeDeviceOpen: Error allocating user buffer memory.";
- goto Exit;
- }
-
- if ( options && options->flags & RTAUDIO_SCHEDULE_REALTIME )
- stream_.callbackInfo.priority = 15;
- else
- stream_.callbackInfo.priority = 0;
-
- ///! TODO: RTAUDIO_MINIMIZE_LATENCY // Provide stream buffers directly to callback
- ///! TODO: RTAUDIO_HOG_DEVICE // Exclusive mode
-
- methodResult = SUCCESS;
-
-Exit:
- //clean up
- SAFE_RELEASE( captureDevices );
- SAFE_RELEASE( renderDevices );
- SAFE_RELEASE( devicePtr );
- CoTaskMemFree( deviceFormat );
-
- // if method failed, close the stream
- if ( methodResult == FAILURE )
- closeStream();
-
- if ( !errorText_.empty() )
- error( errorType );
- return methodResult;
-}
-
-//=============================================================================
-
-DWORD WINAPI RtApiWasapi::runWasapiThread( void* wasapiPtr )
-{
- if ( wasapiPtr )
- ( ( RtApiWasapi* ) wasapiPtr )->wasapiThread();
-
- return 0;
-}
-
-DWORD WINAPI RtApiWasapi::stopWasapiThread( void* wasapiPtr )
-{
- if ( wasapiPtr )
- ( ( RtApiWasapi* ) wasapiPtr )->stopStream();
-
- return 0;
-}
-
-DWORD WINAPI RtApiWasapi::abortWasapiThread( void* wasapiPtr )
-{
- if ( wasapiPtr )
- ( ( RtApiWasapi* ) wasapiPtr )->abortStream();
-
- return 0;
-}
-
-//-----------------------------------------------------------------------------
-
-void RtApiWasapi::wasapiThread()
-{
- // as this is a new thread, we must CoInitialize it
- CoInitialize( NULL );
-
- HRESULT hr;
-
- IAudioClient* captureAudioClient = ( ( WasapiHandle* ) stream_.apiHandle )->captureAudioClient;
- IAudioClient* renderAudioClient = ( ( WasapiHandle* ) stream_.apiHandle )->renderAudioClient;
- IAudioCaptureClient* captureClient = ( ( WasapiHandle* ) stream_.apiHandle )->captureClient;
- IAudioRenderClient* renderClient = ( ( WasapiHandle* ) stream_.apiHandle )->renderClient;
- HANDLE captureEvent = ( ( WasapiHandle* ) stream_.apiHandle )->captureEvent;
- HANDLE renderEvent = ( ( WasapiHandle* ) stream_.apiHandle )->renderEvent;
-
- WAVEFORMATEX* captureFormat = NULL;
- WAVEFORMATEX* renderFormat = NULL;
- float captureSrRatio = 0.0f;
- float renderSrRatio = 0.0f;
- WasapiBuffer captureBuffer;
- WasapiBuffer renderBuffer;
-
- // declare local stream variables
- RtAudioCallback callback = ( RtAudioCallback ) stream_.callbackInfo.callback;
- BYTE* streamBuffer = NULL;
- unsigned long captureFlags = 0;
- unsigned int bufferFrameCount = 0;
- unsigned int numFramesPadding = 0;
- unsigned int convBufferSize = 0;
- bool callbackPushed = false;
- bool callbackPulled = false;
- bool callbackStopped = false;
- int callbackResult = 0;
-
- // convBuffer is used to store converted buffers between WASAPI and the user
- char* convBuffer = NULL;
- unsigned int convBuffSize = 0;
- unsigned int deviceBuffSize = 0;
-
- errorText_.clear();
- RtAudioError::Type errorType = RtAudioError::DRIVER_ERROR;
-
- // Attempt to assign "Pro Audio" characteristic to thread
- HMODULE AvrtDll = LoadLibrary( (LPCTSTR) "AVRT.dll" );
- if ( AvrtDll ) {
- DWORD taskIndex = 0;
- TAvSetMmThreadCharacteristicsPtr AvSetMmThreadCharacteristicsPtr = ( TAvSetMmThreadCharacteristicsPtr ) GetProcAddress( AvrtDll, "AvSetMmThreadCharacteristicsW" );
- AvSetMmThreadCharacteristicsPtr( L"Pro Audio", &taskIndex );
- FreeLibrary( AvrtDll );
- }
-
- // start capture stream if applicable
- if ( captureAudioClient ) {
- hr = captureAudioClient->GetMixFormat( &captureFormat );
- if ( FAILED( hr ) ) {
- errorText_ = "RtApiWasapi::wasapiThread: Unable to retrieve device mix format.";
- goto Exit;
- }
-
- captureSrRatio = ( ( float ) captureFormat->nSamplesPerSec / stream_.sampleRate );
-
- // initialize capture stream according to desire buffer size
- float desiredBufferSize = stream_.bufferSize * captureSrRatio;
- REFERENCE_TIME desiredBufferPeriod = ( REFERENCE_TIME ) ( ( float ) desiredBufferSize * 10000000 / captureFormat->nSamplesPerSec );
-
- if ( !captureClient ) {
- hr = captureAudioClient->Initialize( AUDCLNT_SHAREMODE_SHARED,
- AUDCLNT_STREAMFLAGS_EVENTCALLBACK,
- desiredBufferPeriod,
- desiredBufferPeriod,
- captureFormat,
- NULL );
- if ( FAILED( hr ) ) {
- errorText_ = "RtApiWasapi::wasapiThread: Unable to initialize capture audio client.";
- goto Exit;
- }
-
- hr = captureAudioClient->GetService( __uuidof( IAudioCaptureClient ),
- ( void** ) &captureClient );
- if ( FAILED( hr ) ) {
- errorText_ = "RtApiWasapi::wasapiThread: Unable to retrieve capture client handle.";
- goto Exit;
- }
-
- // configure captureEvent to trigger on every available capture buffer
- captureEvent = CreateEvent( NULL, FALSE, FALSE, NULL );
- if ( !captureEvent ) {
- errorType = RtAudioError::SYSTEM_ERROR;
- errorText_ = "RtApiWasapi::wasapiThread: Unable to create capture event.";
- goto Exit;
- }
-
- hr = captureAudioClient->SetEventHandle( captureEvent );
- if ( FAILED( hr ) ) {
- errorText_ = "RtApiWasapi::wasapiThread: Unable to set capture event handle.";
- goto Exit;
- }
-
- ( ( WasapiHandle* ) stream_.apiHandle )->captureClient = captureClient;
- ( ( WasapiHandle* ) stream_.apiHandle )->captureEvent = captureEvent;
- }
-
- unsigned int inBufferSize = 0;
- hr = captureAudioClient->GetBufferSize( &inBufferSize );
- if ( FAILED( hr ) ) {
- errorText_ = "RtApiWasapi::wasapiThread: Unable to get capture buffer size.";
- goto Exit;
- }
-
- // scale outBufferSize according to stream->user sample rate ratio
- unsigned int outBufferSize = ( unsigned int ) ( stream_.bufferSize * captureSrRatio ) * stream_.nDeviceChannels[INPUT];
- inBufferSize *= stream_.nDeviceChannels[INPUT];
-
- // set captureBuffer size
- captureBuffer.setBufferSize( inBufferSize + outBufferSize, formatBytes( stream_.deviceFormat[INPUT] ) );
-
- // reset the capture stream
- hr = captureAudioClient->Reset();
- if ( FAILED( hr ) ) {
- errorText_ = "RtApiWasapi::wasapiThread: Unable to reset capture stream.";
- goto Exit;
- }
-
- // start the capture stream
- hr = captureAudioClient->Start();
- if ( FAILED( hr ) ) {
- errorText_ = "RtApiWasapi::wasapiThread: Unable to start capture stream.";
- goto Exit;
- }
- }
-
- // start render stream if applicable
- if ( renderAudioClient ) {
- hr = renderAudioClient->GetMixFormat( &renderFormat );
- if ( FAILED( hr ) ) {
- errorText_ = "RtApiWasapi::wasapiThread: Unable to retrieve device mix format.";
- goto Exit;
- }
-
- renderSrRatio = ( ( float ) renderFormat->nSamplesPerSec / stream_.sampleRate );
-
- // initialize render stream according to desire buffer size
- float desiredBufferSize = stream_.bufferSize * renderSrRatio;
- REFERENCE_TIME desiredBufferPeriod = ( REFERENCE_TIME ) ( ( float ) desiredBufferSize * 10000000 / renderFormat->nSamplesPerSec );
-
- if ( !renderClient ) {
- hr = renderAudioClient->Initialize( AUDCLNT_SHAREMODE_SHARED,
- AUDCLNT_STREAMFLAGS_EVENTCALLBACK,
- desiredBufferPeriod,
- desiredBufferPeriod,
- renderFormat,
- NULL );
- if ( FAILED( hr ) ) {
- errorText_ = "RtApiWasapi::wasapiThread: Unable to initialize render audio client.";
- goto Exit;
- }
-
- hr = renderAudioClient->GetService( __uuidof( IAudioRenderClient ),
- ( void** ) &renderClient );
- if ( FAILED( hr ) ) {
- errorText_ = "RtApiWasapi::wasapiThread: Unable to retrieve render client handle.";
- goto Exit;
- }
-
- // configure renderEvent to trigger on every available render buffer
- renderEvent = CreateEvent( NULL, FALSE, FALSE, NULL );
- if ( !renderEvent ) {
- errorType = RtAudioError::SYSTEM_ERROR;
- errorText_ = "RtApiWasapi::wasapiThread: Unable to create render event.";
- goto Exit;
- }
-
- hr = renderAudioClient->SetEventHandle( renderEvent );
- if ( FAILED( hr ) ) {
- errorText_ = "RtApiWasapi::wasapiThread: Unable to set render event handle.";
- goto Exit;
- }
-
- ( ( WasapiHandle* ) stream_.apiHandle )->renderClient = renderClient;
- ( ( WasapiHandle* ) stream_.apiHandle )->renderEvent = renderEvent;
- }
-
- unsigned int outBufferSize = 0;
- hr = renderAudioClient->GetBufferSize( &outBufferSize );
- if ( FAILED( hr ) ) {
- errorText_ = "RtApiWasapi::wasapiThread: Unable to get render buffer size.";
- goto Exit;
- }
-
- // scale inBufferSize according to user->stream sample rate ratio
- unsigned int inBufferSize = ( unsigned int ) ( stream_.bufferSize * renderSrRatio ) * stream_.nDeviceChannels[OUTPUT];
- outBufferSize *= stream_.nDeviceChannels[OUTPUT];
-
- // set renderBuffer size
- renderBuffer.setBufferSize( inBufferSize + outBufferSize, formatBytes( stream_.deviceFormat[OUTPUT] ) );
-
- // reset the render stream
- hr = renderAudioClient->Reset();
- if ( FAILED( hr ) ) {
- errorText_ = "RtApiWasapi::wasapiThread: Unable to reset render stream.";
- goto Exit;
- }
-
- // start the render stream
- hr = renderAudioClient->Start();
- if ( FAILED( hr ) ) {
- errorText_ = "RtApiWasapi::wasapiThread: Unable to start render stream.";
- goto Exit;
- }
- }
-
- if ( stream_.mode == INPUT ) {
- convBuffSize = ( size_t ) ( stream_.bufferSize * captureSrRatio ) * stream_.nDeviceChannels[INPUT] * formatBytes( stream_.deviceFormat[INPUT] );
- deviceBuffSize = stream_.bufferSize * stream_.nDeviceChannels[INPUT] * formatBytes( stream_.deviceFormat[INPUT] );
- }
- else if ( stream_.mode == OUTPUT ) {
- convBuffSize = ( size_t ) ( stream_.bufferSize * renderSrRatio ) * stream_.nDeviceChannels[OUTPUT] * formatBytes( stream_.deviceFormat[OUTPUT] );
- deviceBuffSize = stream_.bufferSize * stream_.nDeviceChannels[OUTPUT] * formatBytes( stream_.deviceFormat[OUTPUT] );
- }
- else if ( stream_.mode == DUPLEX ) {
- convBuffSize = std::max( ( size_t ) ( stream_.bufferSize * captureSrRatio ) * stream_.nDeviceChannels[INPUT] * formatBytes( stream_.deviceFormat[INPUT] ),
- ( size_t ) ( stream_.bufferSize * renderSrRatio ) * stream_.nDeviceChannels[OUTPUT] * formatBytes( stream_.deviceFormat[OUTPUT] ) );
- deviceBuffSize = std::max( stream_.bufferSize * stream_.nDeviceChannels[INPUT] * formatBytes( stream_.deviceFormat[INPUT] ),
- stream_.bufferSize * stream_.nDeviceChannels[OUTPUT] * formatBytes( stream_.deviceFormat[OUTPUT] ) );
- }
-
- convBuffer = ( char* ) malloc( convBuffSize );
- stream_.deviceBuffer = ( char* ) malloc( deviceBuffSize );
- if ( !convBuffer || !stream_.deviceBuffer ) {
- errorType = RtAudioError::MEMORY_ERROR;
- errorText_ = "RtApiWasapi::wasapiThread: Error allocating device buffer memory.";
- goto Exit;
- }
-
- // stream process loop
- while ( stream_.state != STREAM_STOPPING ) {
- if ( !callbackPulled ) {
- // Callback Input
- // ==============
- // 1. Pull callback buffer from inputBuffer
- // 2. If 1. was successful: Convert callback buffer to user sample rate and channel count
- // Convert callback buffer to user format
-
- if ( captureAudioClient ) {
- // Pull callback buffer from inputBuffer
- callbackPulled = captureBuffer.pullBuffer( convBuffer,
- ( unsigned int ) ( stream_.bufferSize * captureSrRatio ) * stream_.nDeviceChannels[INPUT],
- stream_.deviceFormat[INPUT] );
-
- if ( callbackPulled ) {
- // Convert callback buffer to user sample rate
- convertBufferWasapi( stream_.deviceBuffer,
- convBuffer,
- stream_.nDeviceChannels[INPUT],
- captureFormat->nSamplesPerSec,
- stream_.sampleRate,
- ( unsigned int ) ( stream_.bufferSize * captureSrRatio ),
- convBufferSize,
- stream_.deviceFormat[INPUT] );
-
- if ( stream_.doConvertBuffer[INPUT] ) {
- // Convert callback buffer to user format
- convertBuffer( stream_.userBuffer[INPUT],
- stream_.deviceBuffer,
- stream_.convertInfo[INPUT] );
- }
- else {
- // no further conversion, simple copy deviceBuffer to userBuffer
- memcpy( stream_.userBuffer[INPUT],
- stream_.deviceBuffer,
- stream_.bufferSize * stream_.nUserChannels[INPUT] * formatBytes( stream_.userFormat ) );
- }
- }
- }
- else {
- // if there is no capture stream, set callbackPulled flag
- callbackPulled = true;
- }
-
- // Execute Callback
- // ================
- // 1. Execute user callback method
- // 2. Handle return value from callback
-
- // if callback has not requested the stream to stop
- if ( callbackPulled && !callbackStopped ) {
- // Execute user callback method
- callbackResult = callback( stream_.userBuffer[OUTPUT],
- stream_.userBuffer[INPUT],
- stream_.bufferSize,
- getStreamTime(),
- captureFlags & AUDCLNT_BUFFERFLAGS_DATA_DISCONTINUITY ? RTAUDIO_INPUT_OVERFLOW : 0,
- stream_.callbackInfo.userData );
-
- // Handle return value from callback
- if ( callbackResult == 1 ) {
- // instantiate a thread to stop this thread
- HANDLE threadHandle = CreateThread( NULL, 0, stopWasapiThread, this, 0, NULL );
- if ( !threadHandle ) {
- errorType = RtAudioError::THREAD_ERROR;
- errorText_ = "RtApiWasapi::wasapiThread: Unable to instantiate stream stop thread.";
- goto Exit;
- }
- else if ( !CloseHandle( threadHandle ) ) {
- errorType = RtAudioError::THREAD_ERROR;
- errorText_ = "RtApiWasapi::wasapiThread: Unable to close stream stop thread handle.";
- goto Exit;
- }
-
- callbackStopped = true;
- }
- else if ( callbackResult == 2 ) {
- // instantiate a thread to stop this thread
- HANDLE threadHandle = CreateThread( NULL, 0, abortWasapiThread, this, 0, NULL );
- if ( !threadHandle ) {
- errorType = RtAudioError::THREAD_ERROR;
- errorText_ = "RtApiWasapi::wasapiThread: Unable to instantiate stream abort thread.";
- goto Exit;
- }
- else if ( !CloseHandle( threadHandle ) ) {
- errorType = RtAudioError::THREAD_ERROR;
- errorText_ = "RtApiWasapi::wasapiThread: Unable to close stream abort thread handle.";
- goto Exit;
- }
-
- callbackStopped = true;
- }
- }
- }
-
- // Callback Output
- // ===============
- // 1. Convert callback buffer to stream format
- // 2. Convert callback buffer to stream sample rate and channel count
- // 3. Push callback buffer into outputBuffer
-
- if ( renderAudioClient && callbackPulled ) {
- if ( stream_.doConvertBuffer[OUTPUT] ) {
- // Convert callback buffer to stream format
- convertBuffer( stream_.deviceBuffer,
- stream_.userBuffer[OUTPUT],
- stream_.convertInfo[OUTPUT] );
-
- }
-
- // Convert callback buffer to stream sample rate
- convertBufferWasapi( convBuffer,
- stream_.deviceBuffer,
- stream_.nDeviceChannels[OUTPUT],
- stream_.sampleRate,
- renderFormat->nSamplesPerSec,
- stream_.bufferSize,
- convBufferSize,
- stream_.deviceFormat[OUTPUT] );
-
- // Push callback buffer into outputBuffer
- callbackPushed = renderBuffer.pushBuffer( convBuffer,
- convBufferSize * stream_.nDeviceChannels[OUTPUT],
- stream_.deviceFormat[OUTPUT] );
- }
- else {
- // if there is no render stream, set callbackPushed flag
- callbackPushed = true;
- }
-
- // Stream Capture
- // ==============
- // 1. Get capture buffer from stream
- // 2. Push capture buffer into inputBuffer
- // 3. If 2. was successful: Release capture buffer
-
- if ( captureAudioClient ) {
- // if the callback input buffer was not pulled from captureBuffer, wait for next capture event
- if ( !callbackPulled ) {
- WaitForSingleObject( captureEvent, INFINITE );
- }
-
- // Get capture buffer from stream
- hr = captureClient->GetBuffer( &streamBuffer,
- &bufferFrameCount,
- &captureFlags, NULL, NULL );
- if ( FAILED( hr ) ) {
- errorText_ = "RtApiWasapi::wasapiThread: Unable to retrieve capture buffer.";
- goto Exit;
- }
-
- if ( bufferFrameCount != 0 ) {
- // Push capture buffer into inputBuffer
- if ( captureBuffer.pushBuffer( ( char* ) streamBuffer,
- bufferFrameCount * stream_.nDeviceChannels[INPUT],
- stream_.deviceFormat[INPUT] ) )
- {
- // Release capture buffer
- hr = captureClient->ReleaseBuffer( bufferFrameCount );
- if ( FAILED( hr ) ) {
- errorText_ = "RtApiWasapi::wasapiThread: Unable to release capture buffer.";
- goto Exit;
- }
- }
- else
- {
- // Inform WASAPI that capture was unsuccessful
- hr = captureClient->ReleaseBuffer( 0 );
- if ( FAILED( hr ) ) {
- errorText_ = "RtApiWasapi::wasapiThread: Unable to release capture buffer.";
- goto Exit;
- }
- }
- }
- else
- {
- // Inform WASAPI that capture was unsuccessful
- hr = captureClient->ReleaseBuffer( 0 );
- if ( FAILED( hr ) ) {
- errorText_ = "RtApiWasapi::wasapiThread: Unable to release capture buffer.";
- goto Exit;
- }
- }
- }
-
- // Stream Render
- // =============
- // 1. Get render buffer from stream
- // 2. Pull next buffer from outputBuffer
- // 3. If 2. was successful: Fill render buffer with next buffer
- // Release render buffer
-
- if ( renderAudioClient ) {
- // if the callback output buffer was not pushed to renderBuffer, wait for next render event
- if ( callbackPulled && !callbackPushed ) {
- WaitForSingleObject( renderEvent, INFINITE );
- }
-
- // Get render buffer from stream
- hr = renderAudioClient->GetBufferSize( &bufferFrameCount );
- if ( FAILED( hr ) ) {
- errorText_ = "RtApiWasapi::wasapiThread: Unable to retrieve render buffer size.";
- goto Exit;
- }
-
- hr = renderAudioClient->GetCurrentPadding( &numFramesPadding );
- if ( FAILED( hr ) ) {
- errorText_ = "RtApiWasapi::wasapiThread: Unable to retrieve render buffer padding.";
- goto Exit;
- }
-
- bufferFrameCount -= numFramesPadding;
-
- if ( bufferFrameCount != 0 ) {
- hr = renderClient->GetBuffer( bufferFrameCount, &streamBuffer );
- if ( FAILED( hr ) ) {
- errorText_ = "RtApiWasapi::wasapiThread: Unable to retrieve render buffer.";
- goto Exit;
- }
-
- // Pull next buffer from outputBuffer
- // Fill render buffer with next buffer
- if ( renderBuffer.pullBuffer( ( char* ) streamBuffer,
- bufferFrameCount * stream_.nDeviceChannels[OUTPUT],
- stream_.deviceFormat[OUTPUT] ) )
- {
- // Release render buffer
- hr = renderClient->ReleaseBuffer( bufferFrameCount, 0 );
- if ( FAILED( hr ) ) {
- errorText_ = "RtApiWasapi::wasapiThread: Unable to release render buffer.";
- goto Exit;
- }
- }
- else
- {
- // Inform WASAPI that render was unsuccessful
- hr = renderClient->ReleaseBuffer( 0, 0 );
- if ( FAILED( hr ) ) {
- errorText_ = "RtApiWasapi::wasapiThread: Unable to release render buffer.";
- goto Exit;
- }
- }
- }
- else
- {
- // Inform WASAPI that render was unsuccessful
- hr = renderClient->ReleaseBuffer( 0, 0 );
- if ( FAILED( hr ) ) {
- errorText_ = "RtApiWasapi::wasapiThread: Unable to release render buffer.";
- goto Exit;
- }
- }
- }
-
- // if the callback buffer was pushed renderBuffer reset callbackPulled flag
- if ( callbackPushed ) {
- callbackPulled = false;
- // tick stream time
- RtApi::tickStreamTime();
- }
-
- }
-
-Exit:
- // clean up
- CoTaskMemFree( captureFormat );
- CoTaskMemFree( renderFormat );
-
- free ( convBuffer );
-
- CoUninitialize();
-
- // update stream state
- stream_.state = STREAM_STOPPED;
-
- if ( errorText_.empty() )
- return;
- else
- error( errorType );
-}
-
-//******************** End of __WINDOWS_WASAPI__ *********************//
-#endif
-
-
-#if defined(__WINDOWS_DS__) // Windows DirectSound API
-
-// Modified by Robin Davies, October 2005
-// - Improvements to DirectX pointer chasing.
-// - Bug fix for non-power-of-two Asio granularity used by Edirol PCR-A30.
-// - Auto-call CoInitialize for DSOUND and ASIO platforms.
-// Various revisions for RtAudio 4.0 by Gary Scavone, April 2007
-// Changed device query structure for RtAudio 4.0.7, January 2010
-
-#include <dsound.h>
-#include <assert.h>
-#include <algorithm>
-
-#if defined(__MINGW32__)
- // missing from latest mingw winapi
-#define WAVE_FORMAT_96M08 0x00010000 /* 96 kHz, Mono, 8-bit */
-#define WAVE_FORMAT_96S08 0x00020000 /* 96 kHz, Stereo, 8-bit */
-#define WAVE_FORMAT_96M16 0x00040000 /* 96 kHz, Mono, 16-bit */
-#define WAVE_FORMAT_96S16 0x00080000 /* 96 kHz, Stereo, 16-bit */
-#endif
-
-#define MINIMUM_DEVICE_BUFFER_SIZE 32768
-
-#ifdef _MSC_VER // if Microsoft Visual C++
-#pragma comment( lib, "winmm.lib" ) // then, auto-link winmm.lib. Otherwise, it has to be added manually.
-#endif
-
-static inline DWORD dsPointerBetween( DWORD pointer, DWORD laterPointer, DWORD earlierPointer, DWORD bufferSize )
-{
- if ( pointer > bufferSize ) pointer -= bufferSize;
- if ( laterPointer < earlierPointer ) laterPointer += bufferSize;
- if ( pointer < earlierPointer ) pointer += bufferSize;
- return pointer >= earlierPointer && pointer < laterPointer;
-}
-
-// A structure to hold various information related to the DirectSound
-// API implementation.
-struct DsHandle {
- unsigned int drainCounter; // Tracks callback counts when draining
- bool internalDrain; // Indicates if stop is initiated from callback or not.
- void *id[2];
- void *buffer[2];
- bool xrun[2];
- UINT bufferPointer[2];
- DWORD dsBufferSize[2];
- DWORD dsPointerLeadTime[2]; // the number of bytes ahead of the safe pointer to lead by.
- HANDLE condition;
-
- DsHandle()
- :drainCounter(0), internalDrain(false) { id[0] = 0; id[1] = 0; buffer[0] = 0; buffer[1] = 0; xrun[0] = false; xrun[1] = false; bufferPointer[0] = 0; bufferPointer[1] = 0; }
-};
-
-// Declarations for utility functions, callbacks, and structures
-// specific to the DirectSound implementation.
-static BOOL CALLBACK deviceQueryCallback( LPGUID lpguid,
- LPCTSTR description,
- LPCTSTR module,
- LPVOID lpContext );
-
-static const char* getErrorString( int code );
-
-static unsigned __stdcall callbackHandler( void *ptr );
-
-struct DsDevice {
- LPGUID id[2];
- bool validId[2];
- bool found;
- std::string name;
-
- DsDevice()
- : found(false) { validId[0] = false; validId[1] = false; }
-};
-
-struct DsProbeData {
- bool isInput;
- std::vector<struct DsDevice>* dsDevices;
-};
-
-RtApiDs :: RtApiDs()
-{
- // Dsound will run both-threaded. If CoInitialize fails, then just
- // accept whatever the mainline chose for a threading model.
- coInitialized_ = false;
- HRESULT hr = CoInitialize( NULL );
- if ( !FAILED( hr ) ) coInitialized_ = true;
-}
-
-RtApiDs :: ~RtApiDs()
-{
- if ( coInitialized_ ) CoUninitialize(); // balanced call.
- if ( stream_.state != STREAM_CLOSED ) closeStream();
-}
-
-// The DirectSound default output is always the first device.
-unsigned int RtApiDs :: getDefaultOutputDevice( void )
-{
- return 0;
-}
-
-// The DirectSound default input is always the first input device,
-// which is the first capture device enumerated.
-unsigned int RtApiDs :: getDefaultInputDevice( void )
-{
- return 0;
-}
-
-unsigned int RtApiDs :: getDeviceCount( void )
-{
- // Set query flag for previously found devices to false, so that we
- // can check for any devices that have disappeared.
- for ( unsigned int i=0; i<dsDevices.size(); i++ )
- dsDevices[i].found = false;
-
- // Query DirectSound devices.
- struct DsProbeData probeInfo;
- probeInfo.isInput = false;
- probeInfo.dsDevices = &dsDevices;
- HRESULT result = DirectSoundEnumerate( (LPDSENUMCALLBACK) deviceQueryCallback, &probeInfo );
- if ( FAILED( result ) ) {
- errorStream_ << "RtApiDs::getDeviceCount: error (" << getErrorString( result ) << ") enumerating output devices!";
- errorText_ = errorStream_.str();
- error( RtAudioError::WARNING );
- }
-
- // Query DirectSoundCapture devices.
- probeInfo.isInput = true;
- result = DirectSoundCaptureEnumerate( (LPDSENUMCALLBACK) deviceQueryCallback, &probeInfo );
- if ( FAILED( result ) ) {
- errorStream_ << "RtApiDs::getDeviceCount: error (" << getErrorString( result ) << ") enumerating input devices!";
- errorText_ = errorStream_.str();
- error( RtAudioError::WARNING );
- }
-
- // Clean out any devices that may have disappeared (code update submitted by Eli Zehngut).
- for ( unsigned int i=0; i<dsDevices.size(); ) {
- if ( dsDevices[i].found == false ) dsDevices.erase( dsDevices.begin() + i );
- else i++;
- }
-
- return static_cast<unsigned int>(dsDevices.size());
-}
-
-RtAudio::DeviceInfo RtApiDs :: getDeviceInfo( unsigned int device )
-{
- RtAudio::DeviceInfo info;
- info.probed = false;
-
- if ( dsDevices.size() == 0 ) {
- // Force a query of all devices
- getDeviceCount();
- if ( dsDevices.size() == 0 ) {
- errorText_ = "RtApiDs::getDeviceInfo: no devices found!";
- error( RtAudioError::INVALID_USE );
- return info;
- }
- }
-
- if ( device >= dsDevices.size() ) {
- errorText_ = "RtApiDs::getDeviceInfo: device ID is invalid!";
- error( RtAudioError::INVALID_USE );
- return info;
- }
-
- HRESULT result;
- if ( dsDevices[ device ].validId[0] == false ) goto probeInput;
-
- LPDIRECTSOUND output;
- DSCAPS outCaps;
- result = DirectSoundCreate( dsDevices[ device ].id[0], &output, NULL );
- if ( FAILED( result ) ) {
- errorStream_ << "RtApiDs::getDeviceInfo: error (" << getErrorString( result ) << ") opening output device (" << dsDevices[ device ].name << ")!";
- errorText_ = errorStream_.str();
- error( RtAudioError::WARNING );
- goto probeInput;
- }
-
- outCaps.dwSize = sizeof( outCaps );
- result = output->GetCaps( &outCaps );
- if ( FAILED( result ) ) {
- output->Release();
- errorStream_ << "RtApiDs::getDeviceInfo: error (" << getErrorString( result ) << ") getting capabilities!";
- errorText_ = errorStream_.str();
- error( RtAudioError::WARNING );
- goto probeInput;
- }
-
- // Get output channel information.
- info.outputChannels = ( outCaps.dwFlags & DSCAPS_PRIMARYSTEREO ) ? 2 : 1;
-
- // Get sample rate information.
- info.sampleRates.clear();
- for ( unsigned int k=0; k<MAX_SAMPLE_RATES; k++ ) {
- if ( SAMPLE_RATES[k] >= (unsigned int) outCaps.dwMinSecondarySampleRate &&
- SAMPLE_RATES[k] <= (unsigned int) outCaps.dwMaxSecondarySampleRate ) {
- info.sampleRates.push_back( SAMPLE_RATES[k] );
-
- if ( !info.preferredSampleRate || ( SAMPLE_RATES[k] <= 48000 && SAMPLE_RATES[k] > info.preferredSampleRate ) )
- info.preferredSampleRate = SAMPLE_RATES[k];
- }
- }
-
- // Get format information.
- if ( outCaps.dwFlags & DSCAPS_PRIMARY16BIT ) info.nativeFormats |= RTAUDIO_SINT16;
- if ( outCaps.dwFlags & DSCAPS_PRIMARY8BIT ) info.nativeFormats |= RTAUDIO_SINT8;
-
- output->Release();
-
- if ( getDefaultOutputDevice() == device )
- info.isDefaultOutput = true;
-
- if ( dsDevices[ device ].validId[1] == false ) {
- info.name = dsDevices[ device ].name;
- info.probed = true;
- return info;
- }
-
- probeInput:
-
- LPDIRECTSOUNDCAPTURE input;
- result = DirectSoundCaptureCreate( dsDevices[ device ].id[1], &input, NULL );
- if ( FAILED( result ) ) {
- errorStream_ << "RtApiDs::getDeviceInfo: error (" << getErrorString( result ) << ") opening input device (" << dsDevices[ device ].name << ")!";
- errorText_ = errorStream_.str();
- error( RtAudioError::WARNING );
- return info;
- }
-
- DSCCAPS inCaps;
- inCaps.dwSize = sizeof( inCaps );
- result = input->GetCaps( &inCaps );
- if ( FAILED( result ) ) {
- input->Release();
- errorStream_ << "RtApiDs::getDeviceInfo: error (" << getErrorString( result ) << ") getting object capabilities (" << dsDevices[ device ].name << ")!";
- errorText_ = errorStream_.str();
- error( RtAudioError::WARNING );
- return info;
- }
-
- // Get input channel information.
- info.inputChannels = inCaps.dwChannels;
-
- // Get sample rate and format information.
- std::vector<unsigned int> rates;
- if ( inCaps.dwChannels >= 2 ) {
- if ( inCaps.dwFormats & WAVE_FORMAT_1S16 ) info.nativeFormats |= RTAUDIO_SINT16;
- if ( inCaps.dwFormats & WAVE_FORMAT_2S16 ) info.nativeFormats |= RTAUDIO_SINT16;
- if ( inCaps.dwFormats & WAVE_FORMAT_4S16 ) info.nativeFormats |= RTAUDIO_SINT16;
- if ( inCaps.dwFormats & WAVE_FORMAT_96S16 ) info.nativeFormats |= RTAUDIO_SINT16;
- if ( inCaps.dwFormats & WAVE_FORMAT_1S08 ) info.nativeFormats |= RTAUDIO_SINT8;
- if ( inCaps.dwFormats & WAVE_FORMAT_2S08 ) info.nativeFormats |= RTAUDIO_SINT8;
- if ( inCaps.dwFormats & WAVE_FORMAT_4S08 ) info.nativeFormats |= RTAUDIO_SINT8;
- if ( inCaps.dwFormats & WAVE_FORMAT_96S08 ) info.nativeFormats |= RTAUDIO_SINT8;
-
- if ( info.nativeFormats & RTAUDIO_SINT16 ) {
- if ( inCaps.dwFormats & WAVE_FORMAT_1S16 ) rates.push_back( 11025 );
- if ( inCaps.dwFormats & WAVE_FORMAT_2S16 ) rates.push_back( 22050 );
- if ( inCaps.dwFormats & WAVE_FORMAT_4S16 ) rates.push_back( 44100 );
- if ( inCaps.dwFormats & WAVE_FORMAT_96S16 ) rates.push_back( 96000 );
- }
- else if ( info.nativeFormats & RTAUDIO_SINT8 ) {
- if ( inCaps.dwFormats & WAVE_FORMAT_1S08 ) rates.push_back( 11025 );
- if ( inCaps.dwFormats & WAVE_FORMAT_2S08 ) rates.push_back( 22050 );
- if ( inCaps.dwFormats & WAVE_FORMAT_4S08 ) rates.push_back( 44100 );
- if ( inCaps.dwFormats & WAVE_FORMAT_96S08 ) rates.push_back( 96000 );
- }
- }
- else if ( inCaps.dwChannels == 1 ) {
- if ( inCaps.dwFormats & WAVE_FORMAT_1M16 ) info.nativeFormats |= RTAUDIO_SINT16;
- if ( inCaps.dwFormats & WAVE_FORMAT_2M16 ) info.nativeFormats |= RTAUDIO_SINT16;
- if ( inCaps.dwFormats & WAVE_FORMAT_4M16 ) info.nativeFormats |= RTAUDIO_SINT16;
- if ( inCaps.dwFormats & WAVE_FORMAT_96M16 ) info.nativeFormats |= RTAUDIO_SINT16;
- if ( inCaps.dwFormats & WAVE_FORMAT_1M08 ) info.nativeFormats |= RTAUDIO_SINT8;
- if ( inCaps.dwFormats & WAVE_FORMAT_2M08 ) info.nativeFormats |= RTAUDIO_SINT8;
- if ( inCaps.dwFormats & WAVE_FORMAT_4M08 ) info.nativeFormats |= RTAUDIO_SINT8;
- if ( inCaps.dwFormats & WAVE_FORMAT_96M08 ) info.nativeFormats |= RTAUDIO_SINT8;
-
- if ( info.nativeFormats & RTAUDIO_SINT16 ) {
- if ( inCaps.dwFormats & WAVE_FORMAT_1M16 ) rates.push_back( 11025 );
- if ( inCaps.dwFormats & WAVE_FORMAT_2M16 ) rates.push_back( 22050 );
- if ( inCaps.dwFormats & WAVE_FORMAT_4M16 ) rates.push_back( 44100 );
- if ( inCaps.dwFormats & WAVE_FORMAT_96M16 ) rates.push_back( 96000 );
- }
- else if ( info.nativeFormats & RTAUDIO_SINT8 ) {
- if ( inCaps.dwFormats & WAVE_FORMAT_1M08 ) rates.push_back( 11025 );
- if ( inCaps.dwFormats & WAVE_FORMAT_2M08 ) rates.push_back( 22050 );
- if ( inCaps.dwFormats & WAVE_FORMAT_4M08 ) rates.push_back( 44100 );
- if ( inCaps.dwFormats & WAVE_FORMAT_96M08 ) rates.push_back( 96000 );
- }
- }
- else info.inputChannels = 0; // technically, this would be an error
-
- input->Release();
-
- if ( info.inputChannels == 0 ) return info;
-
- // Copy the supported rates to the info structure but avoid duplication.
- bool found;
- for ( unsigned int i=0; i<rates.size(); i++ ) {
- found = false;
- for ( unsigned int j=0; j<info.sampleRates.size(); j++ ) {
- if ( rates[i] == info.sampleRates[j] ) {
- found = true;
- break;
- }
- }
- if ( found == false ) info.sampleRates.push_back( rates[i] );
- }
- std::sort( info.sampleRates.begin(), info.sampleRates.end() );
-
- // If device opens for both playback and capture, we determine the channels.
- if ( info.outputChannels > 0 && info.inputChannels > 0 )
- info.duplexChannels = (info.outputChannels > info.inputChannels) ? info.inputChannels : info.outputChannels;
-
- if ( device == 0 ) info.isDefaultInput = true;
-
- // Copy name and return.
- info.name = dsDevices[ device ].name;
- info.probed = true;
- return info;
-}
-
-bool RtApiDs :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,
- unsigned int firstChannel, unsigned int sampleRate,
- RtAudioFormat format, unsigned int *bufferSize,
- RtAudio::StreamOptions *options )
-{
- if ( channels + firstChannel > 2 ) {
- errorText_ = "RtApiDs::probeDeviceOpen: DirectSound does not support more than 2 channels per device.";
- return FAILURE;
- }
-
- size_t nDevices = dsDevices.size();
- if ( nDevices == 0 ) {
- // This should not happen because a check is made before this function is called.
- errorText_ = "RtApiDs::probeDeviceOpen: no devices found!";
- return FAILURE;
- }
-
- if ( device >= nDevices ) {
- // This should not happen because a check is made before this function is called.
- errorText_ = "RtApiDs::probeDeviceOpen: device ID is invalid!";
- return FAILURE;
- }
-
- if ( mode == OUTPUT ) {
- if ( dsDevices[ device ].validId[0] == false ) {
- errorStream_ << "RtApiDs::probeDeviceOpen: device (" << device << ") does not support output!";
- errorText_ = errorStream_.str();
- return FAILURE;
- }
- }
- else { // mode == INPUT
- if ( dsDevices[ device ].validId[1] == false ) {
- errorStream_ << "RtApiDs::probeDeviceOpen: device (" << device << ") does not support input!";
- errorText_ = errorStream_.str();
- return FAILURE;
- }
- }
-
- // According to a note in PortAudio, using GetDesktopWindow()
- // instead of GetForegroundWindow() is supposed to avoid problems
- // that occur when the application's window is not the foreground
- // window. Also, if the application window closes before the
- // DirectSound buffer, DirectSound can crash. In the past, I had
- // problems when using GetDesktopWindow() but it seems fine now
- // (January 2010). I'll leave it commented here.
- // HWND hWnd = GetForegroundWindow();
- HWND hWnd = GetDesktopWindow();
-
- // Check the numberOfBuffers parameter and limit the lowest value to
- // two. This is a judgement call and a value of two is probably too
- // low for capture, but it should work for playback.
- int nBuffers = 0;
- if ( options ) nBuffers = options->numberOfBuffers;
- if ( options && options->flags & RTAUDIO_MINIMIZE_LATENCY ) nBuffers = 2;
- if ( nBuffers < 2 ) nBuffers = 3;
-
- // Check the lower range of the user-specified buffer size and set
- // (arbitrarily) to a lower bound of 32.
- if ( *bufferSize < 32 ) *bufferSize = 32;
-
- // Create the wave format structure. The data format setting will
- // be determined later.
- WAVEFORMATEX waveFormat;
- ZeroMemory( &waveFormat, sizeof(WAVEFORMATEX) );
- waveFormat.wFormatTag = WAVE_FORMAT_PCM;
- waveFormat.nChannels = channels + firstChannel;
- waveFormat.nSamplesPerSec = (unsigned long) sampleRate;
-
- // Determine the device buffer size. By default, we'll use the value
- // defined above (32K), but we will grow it to make allowances for
- // very large software buffer sizes.
- DWORD dsBufferSize = MINIMUM_DEVICE_BUFFER_SIZE;
- DWORD dsPointerLeadTime = 0;
-
- void *ohandle = 0, *bhandle = 0;
- HRESULT result;
- if ( mode == OUTPUT ) {
-
- LPDIRECTSOUND output;
- result = DirectSoundCreate( dsDevices[ device ].id[0], &output, NULL );
- if ( FAILED( result ) ) {
- errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") opening output device (" << dsDevices[ device ].name << ")!";
- errorText_ = errorStream_.str();
- return FAILURE;
- }
-
- DSCAPS outCaps;
- outCaps.dwSize = sizeof( outCaps );
- result = output->GetCaps( &outCaps );
- if ( FAILED( result ) ) {
- output->Release();
- errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") getting capabilities (" << dsDevices[ device ].name << ")!";
- errorText_ = errorStream_.str();
- return FAILURE;
- }
-
- // Check channel information.
- if ( channels + firstChannel == 2 && !( outCaps.dwFlags & DSCAPS_PRIMARYSTEREO ) ) {
- errorStream_ << "RtApiDs::getDeviceInfo: the output device (" << dsDevices[ device ].name << ") does not support stereo playback.";
- errorText_ = errorStream_.str();
- return FAILURE;
- }
-
- // Check format information. Use 16-bit format unless not
- // supported or user requests 8-bit.
- if ( outCaps.dwFlags & DSCAPS_PRIMARY16BIT &&
- !( format == RTAUDIO_SINT8 && outCaps.dwFlags & DSCAPS_PRIMARY8BIT ) ) {
- waveFormat.wBitsPerSample = 16;
- stream_.deviceFormat[mode] = RTAUDIO_SINT16;
- }
- else {
- waveFormat.wBitsPerSample = 8;
- stream_.deviceFormat[mode] = RTAUDIO_SINT8;
- }
- stream_.userFormat = format;
-
- // Update wave format structure and buffer information.
- waveFormat.nBlockAlign = waveFormat.nChannels * waveFormat.wBitsPerSample / 8;
- waveFormat.nAvgBytesPerSec = waveFormat.nSamplesPerSec * waveFormat.nBlockAlign;
- dsPointerLeadTime = nBuffers * (*bufferSize) * (waveFormat.wBitsPerSample / 8) * channels;
-
- // If the user wants an even bigger buffer, increase the device buffer size accordingly.
- while ( dsPointerLeadTime * 2U > dsBufferSize )
- dsBufferSize *= 2;
-
- // Set cooperative level to DSSCL_EXCLUSIVE ... sound stops when window focus changes.
- // result = output->SetCooperativeLevel( hWnd, DSSCL_EXCLUSIVE );
- // Set cooperative level to DSSCL_PRIORITY ... sound remains when window focus changes.
- result = output->SetCooperativeLevel( hWnd, DSSCL_PRIORITY );
- if ( FAILED( result ) ) {
- output->Release();
- errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") setting cooperative level (" << dsDevices[ device ].name << ")!";
- errorText_ = errorStream_.str();
- return FAILURE;
- }
-
- // Even though we will write to the secondary buffer, we need to
- // access the primary buffer to set the correct output format
- // (since the default is 8-bit, 22 kHz!). Setup the DS primary
- // buffer description.
- DSBUFFERDESC bufferDescription;
- ZeroMemory( &bufferDescription, sizeof( DSBUFFERDESC ) );
- bufferDescription.dwSize = sizeof( DSBUFFERDESC );
- bufferDescription.dwFlags = DSBCAPS_PRIMARYBUFFER;
-
- // Obtain the primary buffer
- LPDIRECTSOUNDBUFFER buffer;
- result = output->CreateSoundBuffer( &bufferDescription, &buffer, NULL );
- if ( FAILED( result ) ) {
- output->Release();
- errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") accessing primary buffer (" << dsDevices[ device ].name << ")!";
- errorText_ = errorStream_.str();
- return FAILURE;
- }
-
- // Set the primary DS buffer sound format.
- result = buffer->SetFormat( &waveFormat );
- if ( FAILED( result ) ) {
- output->Release();
- errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") setting primary buffer format (" << dsDevices[ device ].name << ")!";
- errorText_ = errorStream_.str();
- return FAILURE;
- }
-
- // Setup the secondary DS buffer description.
- ZeroMemory( &bufferDescription, sizeof( DSBUFFERDESC ) );
- bufferDescription.dwSize = sizeof( DSBUFFERDESC );
- bufferDescription.dwFlags = ( DSBCAPS_STICKYFOCUS |
- DSBCAPS_GLOBALFOCUS |
- DSBCAPS_GETCURRENTPOSITION2 |
- DSBCAPS_LOCHARDWARE ); // Force hardware mixing
- bufferDescription.dwBufferBytes = dsBufferSize;
- bufferDescription.lpwfxFormat = &waveFormat;
-
- // Try to create the secondary DS buffer. If that doesn't work,
- // try to use software mixing. Otherwise, there's a problem.
- result = output->CreateSoundBuffer( &bufferDescription, &buffer, NULL );
- if ( FAILED( result ) ) {
- bufferDescription.dwFlags = ( DSBCAPS_STICKYFOCUS |
- DSBCAPS_GLOBALFOCUS |
- DSBCAPS_GETCURRENTPOSITION2 |
- DSBCAPS_LOCSOFTWARE ); // Force software mixing
- result = output->CreateSoundBuffer( &bufferDescription, &buffer, NULL );
- if ( FAILED( result ) ) {
- output->Release();
- errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") creating secondary buffer (" << dsDevices[ device ].name << ")!";
- errorText_ = errorStream_.str();
- return FAILURE;
- }
- }
-
- // Get the buffer size ... might be different from what we specified.
- DSBCAPS dsbcaps;
- dsbcaps.dwSize = sizeof( DSBCAPS );
- result = buffer->GetCaps( &dsbcaps );
- if ( FAILED( result ) ) {
- output->Release();
- buffer->Release();
- errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") getting buffer settings (" << dsDevices[ device ].name << ")!";
- errorText_ = errorStream_.str();
- return FAILURE;
- }
-
- dsBufferSize = dsbcaps.dwBufferBytes;
-
- // Lock the DS buffer
- LPVOID audioPtr;
- DWORD dataLen;
- result = buffer->Lock( 0, dsBufferSize, &audioPtr, &dataLen, NULL, NULL, 0 );
- if ( FAILED( result ) ) {
- output->Release();
- buffer->Release();
- errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") locking buffer (" << dsDevices[ device ].name << ")!";
- errorText_ = errorStream_.str();
- return FAILURE;
- }
-
- // Zero the DS buffer
- ZeroMemory( audioPtr, dataLen );
-
- // Unlock the DS buffer
- result = buffer->Unlock( audioPtr, dataLen, NULL, 0 );
- if ( FAILED( result ) ) {
- output->Release();
- buffer->Release();
- errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") unlocking buffer (" << dsDevices[ device ].name << ")!";
- errorText_ = errorStream_.str();
- return FAILURE;
- }
-
- ohandle = (void *) output;
- bhandle = (void *) buffer;
- }
-
- if ( mode == INPUT ) {
-
- LPDIRECTSOUNDCAPTURE input;
- result = DirectSoundCaptureCreate( dsDevices[ device ].id[1], &input, NULL );
- if ( FAILED( result ) ) {
- errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") opening input device (" << dsDevices[ device ].name << ")!";
- errorText_ = errorStream_.str();
- return FAILURE;
- }
-
- DSCCAPS inCaps;
- inCaps.dwSize = sizeof( inCaps );
- result = input->GetCaps( &inCaps );
- if ( FAILED( result ) ) {
- input->Release();
- errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") getting input capabilities (" << dsDevices[ device ].name << ")!";
- errorText_ = errorStream_.str();
- return FAILURE;
- }
-
- // Check channel information.
- if ( inCaps.dwChannels < channels + firstChannel ) {
- errorText_ = "RtApiDs::getDeviceInfo: the input device does not support requested input channels.";
- return FAILURE;
- }
-
- // Check format information. Use 16-bit format unless user
- // requests 8-bit.
- DWORD deviceFormats;
- if ( channels + firstChannel == 2 ) {
- deviceFormats = WAVE_FORMAT_1S08 | WAVE_FORMAT_2S08 | WAVE_FORMAT_4S08 | WAVE_FORMAT_96S08;
- if ( format == RTAUDIO_SINT8 && inCaps.dwFormats & deviceFormats ) {
- waveFormat.wBitsPerSample = 8;
- stream_.deviceFormat[mode] = RTAUDIO_SINT8;
- }
- else { // assume 16-bit is supported
- waveFormat.wBitsPerSample = 16;
- stream_.deviceFormat[mode] = RTAUDIO_SINT16;
- }
- }
- else { // channel == 1
- deviceFormats = WAVE_FORMAT_1M08 | WAVE_FORMAT_2M08 | WAVE_FORMAT_4M08 | WAVE_FORMAT_96M08;
- if ( format == RTAUDIO_SINT8 && inCaps.dwFormats & deviceFormats ) {
- waveFormat.wBitsPerSample = 8;
- stream_.deviceFormat[mode] = RTAUDIO_SINT8;
- }
- else { // assume 16-bit is supported
- waveFormat.wBitsPerSample = 16;
- stream_.deviceFormat[mode] = RTAUDIO_SINT16;
- }
- }
- stream_.userFormat = format;
-
- // Update wave format structure and buffer information.
- waveFormat.nBlockAlign = waveFormat.nChannels * waveFormat.wBitsPerSample / 8;
- waveFormat.nAvgBytesPerSec = waveFormat.nSamplesPerSec * waveFormat.nBlockAlign;
- dsPointerLeadTime = nBuffers * (*bufferSize) * (waveFormat.wBitsPerSample / 8) * channels;
-
- // If the user wants an even bigger buffer, increase the device buffer size accordingly.
- while ( dsPointerLeadTime * 2U > dsBufferSize )
- dsBufferSize *= 2;
-
- // Setup the secondary DS buffer description.
- DSCBUFFERDESC bufferDescription;
- ZeroMemory( &bufferDescription, sizeof( DSCBUFFERDESC ) );
- bufferDescription.dwSize = sizeof( DSCBUFFERDESC );
- bufferDescription.dwFlags = 0;
- bufferDescription.dwReserved = 0;
- bufferDescription.dwBufferBytes = dsBufferSize;
- bufferDescription.lpwfxFormat = &waveFormat;
-
- // Create the capture buffer.
- LPDIRECTSOUNDCAPTUREBUFFER buffer;
- result = input->CreateCaptureBuffer( &bufferDescription, &buffer, NULL );
- if ( FAILED( result ) ) {
- input->Release();
- errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") creating input buffer (" << dsDevices[ device ].name << ")!";
- errorText_ = errorStream_.str();
- return FAILURE;
- }
-
- // Get the buffer size ... might be different from what we specified.
- DSCBCAPS dscbcaps;
- dscbcaps.dwSize = sizeof( DSCBCAPS );
- result = buffer->GetCaps( &dscbcaps );
- if ( FAILED( result ) ) {
- input->Release();
- buffer->Release();
- errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") getting buffer settings (" << dsDevices[ device ].name << ")!";
- errorText_ = errorStream_.str();
- return FAILURE;
- }
-
- dsBufferSize = dscbcaps.dwBufferBytes;
-
- // NOTE: We could have a problem here if this is a duplex stream
- // and the play and capture hardware buffer sizes are different
- // (I'm actually not sure if that is a problem or not).
- // Currently, we are not verifying that.
-
- // Lock the capture buffer
- LPVOID audioPtr;
- DWORD dataLen;
- result = buffer->Lock( 0, dsBufferSize, &audioPtr, &dataLen, NULL, NULL, 0 );
- if ( FAILED( result ) ) {
- input->Release();
- buffer->Release();
- errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") locking input buffer (" << dsDevices[ device ].name << ")!";
- errorText_ = errorStream_.str();
- return FAILURE;
- }
-
- // Zero the buffer
- ZeroMemory( audioPtr, dataLen );
-
- // Unlock the buffer
- result = buffer->Unlock( audioPtr, dataLen, NULL, 0 );
- if ( FAILED( result ) ) {
- input->Release();
- buffer->Release();
- errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") unlocking input buffer (" << dsDevices[ device ].name << ")!";
- errorText_ = errorStream_.str();
- return FAILURE;
- }
-
- ohandle = (void *) input;
- bhandle = (void *) buffer;
- }
-
- // Set various stream parameters
- DsHandle *handle = 0;
- stream_.nDeviceChannels[mode] = channels + firstChannel;
- stream_.nUserChannels[mode] = channels;
- stream_.bufferSize = *bufferSize;
- stream_.channelOffset[mode] = firstChannel;
- stream_.deviceInterleaved[mode] = true;
- if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) stream_.userInterleaved = false;
- else stream_.userInterleaved = true;
-
- // Set flag for buffer conversion
- stream_.doConvertBuffer[mode] = false;
- if (stream_.nUserChannels[mode] != stream_.nDeviceChannels[mode])
- stream_.doConvertBuffer[mode] = true;
- if (stream_.userFormat != stream_.deviceFormat[mode])
- stream_.doConvertBuffer[mode] = true;
- if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] &&
- stream_.nUserChannels[mode] > 1 )
- stream_.doConvertBuffer[mode] = true;
-
- // Allocate necessary internal buffers
- long bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat );
- stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 );
- if ( stream_.userBuffer[mode] == NULL ) {
- errorText_ = "RtApiDs::probeDeviceOpen: error allocating user buffer memory.";
- goto error;
- }
-
- if ( stream_.doConvertBuffer[mode] ) {
-
- bool makeBuffer = true;
- bufferBytes = stream_.nDeviceChannels[mode] * formatBytes( stream_.deviceFormat[mode] );
- if ( mode == INPUT ) {
- if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) {
- unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] );
- if ( bufferBytes <= (long) bytesOut ) makeBuffer = false;
- }
- }
-
- if ( makeBuffer ) {
- bufferBytes *= *bufferSize;
- if ( stream_.deviceBuffer ) free( stream_.deviceBuffer );
- stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 );
- if ( stream_.deviceBuffer == NULL ) {
- errorText_ = "RtApiDs::probeDeviceOpen: error allocating device buffer memory.";
- goto error;
- }
- }
- }
-
- // Allocate our DsHandle structures for the stream.
- if ( stream_.apiHandle == 0 ) {
- try {
- handle = new DsHandle;
- }
- catch ( std::bad_alloc& ) {
- errorText_ = "RtApiDs::probeDeviceOpen: error allocating AsioHandle memory.";
- goto error;
- }
-
- // Create a manual-reset event.
- handle->condition = CreateEvent( NULL, // no security
- TRUE, // manual-reset
- FALSE, // non-signaled initially
- NULL ); // unnamed
- stream_.apiHandle = (void *) handle;
- }
- else
- handle = (DsHandle *) stream_.apiHandle;
- handle->id[mode] = ohandle;
- handle->buffer[mode] = bhandle;
- handle->dsBufferSize[mode] = dsBufferSize;
- handle->dsPointerLeadTime[mode] = dsPointerLeadTime;
-
- stream_.device[mode] = device;
- stream_.state = STREAM_STOPPED;
- if ( stream_.mode == OUTPUT && mode == INPUT )
- // We had already set up an output stream.
- stream_.mode = DUPLEX;
- else
- stream_.mode = mode;
- stream_.nBuffers = nBuffers;
- stream_.sampleRate = sampleRate;
-
- // Setup the buffer conversion information structure.
- if ( stream_.doConvertBuffer[mode] ) setConvertInfo( mode, firstChannel );
-
- // Setup the callback thread.
- if ( stream_.callbackInfo.isRunning == false ) {
- unsigned threadId;
- stream_.callbackInfo.isRunning = true;
- stream_.callbackInfo.object = (void *) this;
- stream_.callbackInfo.thread = _beginthreadex( NULL, 0, &callbackHandler,
- &stream_.callbackInfo, 0, &threadId );
- if ( stream_.callbackInfo.thread == 0 ) {
- errorText_ = "RtApiDs::probeDeviceOpen: error creating callback thread!";
- goto error;
- }
-
- // Boost DS thread priority
- SetThreadPriority( (HANDLE) stream_.callbackInfo.thread, THREAD_PRIORITY_HIGHEST );
- }
- return SUCCESS;
-
- error:
- if ( handle ) {
- if ( handle->buffer[0] ) { // the object pointer can be NULL and valid
- LPDIRECTSOUND object = (LPDIRECTSOUND) handle->id[0];
- LPDIRECTSOUNDBUFFER buffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0];
- if ( buffer ) buffer->Release();
- object->Release();
- }
- if ( handle->buffer[1] ) {
- LPDIRECTSOUNDCAPTURE object = (LPDIRECTSOUNDCAPTURE) handle->id[1];
- LPDIRECTSOUNDCAPTUREBUFFER buffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1];
- if ( buffer ) buffer->Release();
- object->Release();
- }
- CloseHandle( handle->condition );
- delete handle;
- stream_.apiHandle = 0;
- }
-
- for ( int i=0; i<2; i++ ) {
- if ( stream_.userBuffer[i] ) {
- free( stream_.userBuffer[i] );
- stream_.userBuffer[i] = 0;
- }
- }
-
- if ( stream_.deviceBuffer ) {
- free( stream_.deviceBuffer );
- stream_.deviceBuffer = 0;
- }
-
- stream_.state = STREAM_CLOSED;
- return FAILURE;
-}
-
-void RtApiDs :: closeStream()
-{
- if ( stream_.state == STREAM_CLOSED ) {
- errorText_ = "RtApiDs::closeStream(): no open stream to close!";
- error( RtAudioError::WARNING );
- return;
- }
-
- // Stop the callback thread.
- stream_.callbackInfo.isRunning = false;
- WaitForSingleObject( (HANDLE) stream_.callbackInfo.thread, INFINITE );
- CloseHandle( (HANDLE) stream_.callbackInfo.thread );
-
- DsHandle *handle = (DsHandle *) stream_.apiHandle;
- if ( handle ) {
- if ( handle->buffer[0] ) { // the object pointer can be NULL and valid
- LPDIRECTSOUND object = (LPDIRECTSOUND) handle->id[0];
- LPDIRECTSOUNDBUFFER buffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0];
- if ( buffer ) {
- buffer->Stop();
- buffer->Release();
- }
- object->Release();
- }
- if ( handle->buffer[1] ) {
- LPDIRECTSOUNDCAPTURE object = (LPDIRECTSOUNDCAPTURE) handle->id[1];
- LPDIRECTSOUNDCAPTUREBUFFER buffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1];
- if ( buffer ) {
- buffer->Stop();
- buffer->Release();
- }
- object->Release();
- }
- CloseHandle( handle->condition );
- delete handle;
- stream_.apiHandle = 0;
- }
-
- for ( int i=0; i<2; i++ ) {
- if ( stream_.userBuffer[i] ) {
- free( stream_.userBuffer[i] );
- stream_.userBuffer[i] = 0;
- }
- }
-
- if ( stream_.deviceBuffer ) {
- free( stream_.deviceBuffer );
- stream_.deviceBuffer = 0;
- }
-
- stream_.mode = UNINITIALIZED;
- stream_.state = STREAM_CLOSED;
-}
-
-void RtApiDs :: startStream()
-{
- verifyStream();
- if ( stream_.state == STREAM_RUNNING ) {
- errorText_ = "RtApiDs::startStream(): the stream is already running!";
- error( RtAudioError::WARNING );
- return;
- }
-
- DsHandle *handle = (DsHandle *) stream_.apiHandle;
-
- // Increase scheduler frequency on lesser windows (a side-effect of
- // increasing timer accuracy). On greater windows (Win2K or later),
- // this is already in effect.
- timeBeginPeriod( 1 );
-
- buffersRolling = false;
- duplexPrerollBytes = 0;
-
- if ( stream_.mode == DUPLEX ) {
- // 0.5 seconds of silence in DUPLEX mode while the devices spin up and synchronize.
- duplexPrerollBytes = (int) ( 0.5 * stream_.sampleRate * formatBytes( stream_.deviceFormat[1] ) * stream_.nDeviceChannels[1] );
- }
-
- HRESULT result = 0;
- if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
-
- LPDIRECTSOUNDBUFFER buffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0];
- result = buffer->Play( 0, 0, DSBPLAY_LOOPING );
- if ( FAILED( result ) ) {
- errorStream_ << "RtApiDs::startStream: error (" << getErrorString( result ) << ") starting output buffer!";
- errorText_ = errorStream_.str();
- goto unlock;
- }
- }
-
- if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {
-
- LPDIRECTSOUNDCAPTUREBUFFER buffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1];
- result = buffer->Start( DSCBSTART_LOOPING );
- if ( FAILED( result ) ) {
- errorStream_ << "RtApiDs::startStream: error (" << getErrorString( result ) << ") starting input buffer!";
- errorText_ = errorStream_.str();
- goto unlock;
- }
- }
-
- handle->drainCounter = 0;
- handle->internalDrain = false;
- ResetEvent( handle->condition );
- stream_.state = STREAM_RUNNING;
-
- unlock:
- if ( FAILED( result ) ) error( RtAudioError::SYSTEM_ERROR );
-}
-
-void RtApiDs :: stopStream()
-{
- verifyStream();
- if ( stream_.state == STREAM_STOPPED ) {
- errorText_ = "RtApiDs::stopStream(): the stream is already stopped!";
- error( RtAudioError::WARNING );
- return;
- }
-
- HRESULT result = 0;
- LPVOID audioPtr;
- DWORD dataLen;
- DsHandle *handle = (DsHandle *) stream_.apiHandle;
- if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
- if ( handle->drainCounter == 0 ) {
- handle->drainCounter = 2;
- WaitForSingleObject( handle->condition, INFINITE ); // block until signaled
- }
-
- stream_.state = STREAM_STOPPED;
-
- MUTEX_LOCK( &stream_.mutex );
-
- // Stop the buffer and clear memory
- LPDIRECTSOUNDBUFFER buffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0];
- result = buffer->Stop();
- if ( FAILED( result ) ) {
- errorStream_ << "RtApiDs::stopStream: error (" << getErrorString( result ) << ") stopping output buffer!";
- errorText_ = errorStream_.str();
- goto unlock;
- }
-
- // Lock the buffer and clear it so that if we start to play again,
- // we won't have old data playing.
- result = buffer->Lock( 0, handle->dsBufferSize[0], &audioPtr, &dataLen, NULL, NULL, 0 );
- if ( FAILED( result ) ) {
- errorStream_ << "RtApiDs::stopStream: error (" << getErrorString( result ) << ") locking output buffer!";
- errorText_ = errorStream_.str();
- goto unlock;
- }
-
- // Zero the DS buffer
- ZeroMemory( audioPtr, dataLen );
-
- // Unlock the DS buffer
- result = buffer->Unlock( audioPtr, dataLen, NULL, 0 );
- if ( FAILED( result ) ) {
- errorStream_ << "RtApiDs::stopStream: error (" << getErrorString( result ) << ") unlocking output buffer!";
- errorText_ = errorStream_.str();
- goto unlock;
- }
-
- // If we start playing again, we must begin at beginning of buffer.
- handle->bufferPointer[0] = 0;
- }
-
- if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {
- LPDIRECTSOUNDCAPTUREBUFFER buffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1];
- audioPtr = NULL;
- dataLen = 0;
-
- stream_.state = STREAM_STOPPED;
-
- if ( stream_.mode != DUPLEX )
- MUTEX_LOCK( &stream_.mutex );
-
- result = buffer->Stop();
- if ( FAILED( result ) ) {
- errorStream_ << "RtApiDs::stopStream: error (" << getErrorString( result ) << ") stopping input buffer!";
- errorText_ = errorStream_.str();
- goto unlock;
- }
-
- // Lock the buffer and clear it so that if we start to play again,
- // we won't have old data playing.
- result = buffer->Lock( 0, handle->dsBufferSize[1], &audioPtr, &dataLen, NULL, NULL, 0 );
- if ( FAILED( result ) ) {
- errorStream_ << "RtApiDs::stopStream: error (" << getErrorString( result ) << ") locking input buffer!";
- errorText_ = errorStream_.str();
- goto unlock;
- }
-
- // Zero the DS buffer
- ZeroMemory( audioPtr, dataLen );
-
- // Unlock the DS buffer
- result = buffer->Unlock( audioPtr, dataLen, NULL, 0 );
- if ( FAILED( result ) ) {
- errorStream_ << "RtApiDs::stopStream: error (" << getErrorString( result ) << ") unlocking input buffer!";
- errorText_ = errorStream_.str();
- goto unlock;
- }
-
- // If we start recording again, we must begin at beginning of buffer.
- handle->bufferPointer[1] = 0;
- }
-
- unlock:
- timeEndPeriod( 1 ); // revert to normal scheduler frequency on lesser windows.
- MUTEX_UNLOCK( &stream_.mutex );
-
- if ( FAILED( result ) ) error( RtAudioError::SYSTEM_ERROR );
-}
-
-void RtApiDs :: abortStream()
-{
- verifyStream();
- if ( stream_.state == STREAM_STOPPED ) {
- errorText_ = "RtApiDs::abortStream(): the stream is already stopped!";
- error( RtAudioError::WARNING );
- return;
- }
-
- DsHandle *handle = (DsHandle *) stream_.apiHandle;
- handle->drainCounter = 2;
-
- stopStream();
-}
-
-void RtApiDs :: callbackEvent()
-{
- if ( stream_.state == STREAM_STOPPED || stream_.state == STREAM_STOPPING ) {
- Sleep( 50 ); // sleep 50 milliseconds
- return;
- }
-
- if ( stream_.state == STREAM_CLOSED ) {
- errorText_ = "RtApiDs::callbackEvent(): the stream is closed ... this shouldn't happen!";
- error( RtAudioError::WARNING );
- return;
- }
-
- CallbackInfo *info = (CallbackInfo *) &stream_.callbackInfo;
- DsHandle *handle = (DsHandle *) stream_.apiHandle;
-
- // Check if we were draining the stream and signal is finished.
- if ( handle->drainCounter > stream_.nBuffers + 2 ) {
-
- stream_.state = STREAM_STOPPING;
- if ( handle->internalDrain == false )
- SetEvent( handle->condition );
- else
- stopStream();
- return;
- }
-
- // Invoke user callback to get fresh output data UNLESS we are
- // draining stream.
- if ( handle->drainCounter == 0 ) {
- RtAudioCallback callback = (RtAudioCallback) info->callback;
- double streamTime = getStreamTime();
- RtAudioStreamStatus status = 0;
- if ( stream_.mode != INPUT && handle->xrun[0] == true ) {
- status |= RTAUDIO_OUTPUT_UNDERFLOW;
- handle->xrun[0] = false;
- }
- if ( stream_.mode != OUTPUT && handle->xrun[1] == true ) {
- status |= RTAUDIO_INPUT_OVERFLOW;
- handle->xrun[1] = false;
- }
- int cbReturnValue = callback( stream_.userBuffer[0], stream_.userBuffer[1],
- stream_.bufferSize, streamTime, status, info->userData );
- if ( cbReturnValue == 2 ) {
- stream_.state = STREAM_STOPPING;
- handle->drainCounter = 2;
- abortStream();
- return;
- }
- else if ( cbReturnValue == 1 ) {
- handle->drainCounter = 1;
- handle->internalDrain = true;
- }
- }
-
- HRESULT result;
- DWORD currentWritePointer, safeWritePointer;
- DWORD currentReadPointer, safeReadPointer;
- UINT nextWritePointer;
-
- LPVOID buffer1 = NULL;
- LPVOID buffer2 = NULL;
- DWORD bufferSize1 = 0;
- DWORD bufferSize2 = 0;
-
- char *buffer;
- long bufferBytes;
-
- MUTEX_LOCK( &stream_.mutex );
- if ( stream_.state == STREAM_STOPPED ) {
- MUTEX_UNLOCK( &stream_.mutex );
- return;
- }
-
- if ( buffersRolling == false ) {
- if ( stream_.mode == DUPLEX ) {
- //assert( handle->dsBufferSize[0] == handle->dsBufferSize[1] );
-
- // It takes a while for the devices to get rolling. As a result,
- // there's no guarantee that the capture and write device pointers
- // will move in lockstep. Wait here for both devices to start
- // rolling, and then set our buffer pointers accordingly.
- // e.g. Crystal Drivers: the capture buffer starts up 5700 to 9600
- // bytes later than the write buffer.
-
- // Stub: a serious risk of having a pre-emptive scheduling round
- // take place between the two GetCurrentPosition calls... but I'm
- // really not sure how to solve the problem. Temporarily boost to
- // Realtime priority, maybe; but I'm not sure what priority the
- // DirectSound service threads run at. We *should* be roughly
- // within a ms or so of correct.
-
- LPDIRECTSOUNDBUFFER dsWriteBuffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0];
- LPDIRECTSOUNDCAPTUREBUFFER dsCaptureBuffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1];
-
- DWORD startSafeWritePointer, startSafeReadPointer;
-
- result = dsWriteBuffer->GetCurrentPosition( NULL, &startSafeWritePointer );
- if ( FAILED( result ) ) {
- errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current write position!";
- errorText_ = errorStream_.str();
- MUTEX_UNLOCK( &stream_.mutex );
- error( RtAudioError::SYSTEM_ERROR );
- return;
- }
- result = dsCaptureBuffer->GetCurrentPosition( NULL, &startSafeReadPointer );
- if ( FAILED( result ) ) {
- errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current read position!";
- errorText_ = errorStream_.str();
- MUTEX_UNLOCK( &stream_.mutex );
- error( RtAudioError::SYSTEM_ERROR );
- return;
- }
- while ( true ) {
- result = dsWriteBuffer->GetCurrentPosition( NULL, &safeWritePointer );
- if ( FAILED( result ) ) {
- errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current write position!";
- errorText_ = errorStream_.str();
- MUTEX_UNLOCK( &stream_.mutex );
- error( RtAudioError::SYSTEM_ERROR );
- return;
- }
- result = dsCaptureBuffer->GetCurrentPosition( NULL, &safeReadPointer );
- if ( FAILED( result ) ) {
- errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current read position!";
- errorText_ = errorStream_.str();
- MUTEX_UNLOCK( &stream_.mutex );
- error( RtAudioError::SYSTEM_ERROR );
- return;
- }
- if ( safeWritePointer != startSafeWritePointer && safeReadPointer != startSafeReadPointer ) break;
- Sleep( 1 );
- }
-
- //assert( handle->dsBufferSize[0] == handle->dsBufferSize[1] );
-
- handle->bufferPointer[0] = safeWritePointer + handle->dsPointerLeadTime[0];
- if ( handle->bufferPointer[0] >= handle->dsBufferSize[0] ) handle->bufferPointer[0] -= handle->dsBufferSize[0];
- handle->bufferPointer[1] = safeReadPointer;
- }
- else if ( stream_.mode == OUTPUT ) {
-
- // Set the proper nextWritePosition after initial startup.
- LPDIRECTSOUNDBUFFER dsWriteBuffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0];
- result = dsWriteBuffer->GetCurrentPosition( &currentWritePointer, &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( &currentWritePointer, &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( &currentReadPointer, &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 aggressive.
- 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( &currentReadPointer, &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( &params );
-
- // 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 occurred.
- 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 occurred.
- 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;
- if ( mask & AFMT_FLOAT )
- info.nativeFormats |= RTAUDIO_FLOAT32;
- if ( mask & AFMT_S24_LE || mask & AFMT_S24_BE )
- info.nativeFormats |= RTAUDIO_SINT24;
-
- // Check that we have at least one supported format
- if ( info.nativeFormats == 0 ) {
- errorStream_ << "RtApiOss::getDeviceInfo: device (" << ainfo.name << ") data format not supported by RtAudio.";
- errorText_ = errorStream_.str();
- error( RtAudioError::WARNING );
- return info;
- }
-
- // Probe the supported sample rates.
- info.sampleRates.clear();
- if ( ainfo.nrates ) {
- for ( unsigned int i=0; i<ainfo.nrates; i++ ) {
- for ( unsigned int k=0; k<MAX_SAMPLE_RATES; k++ ) {
- if ( ainfo.rates[i] == SAMPLE_RATES[k] ) {
- info.sampleRates.push_back( SAMPLE_RATES[k] );
-
- if ( !info.preferredSampleRate || ( SAMPLE_RATES[k] <= 48000 && SAMPLE_RATES[k] > info.preferredSampleRate ) )
- info.preferredSampleRate = SAMPLE_RATES[k];
-
- break;
- }
- }
- }
- }
- else {
- // Check min and max rate values;
- for ( unsigned int k=0; k<MAX_SAMPLE_RATES; k++ ) {
- if ( ainfo.min_rate <= (int) SAMPLE_RATES[k] && ainfo.max_rate >= (int) SAMPLE_RATES[k] ) {
- info.sampleRates.push_back( SAMPLE_RATES[k] );
-
- if ( !info.preferredSampleRate || ( SAMPLE_RATES[k] <= 48000 && SAMPLE_RATES[k] > info.preferredSampleRate ) )
- info.preferredSampleRate = SAMPLE_RATES[k];
- }
- }
- }
-
- if ( info.sampleRates.size() == 0 ) {
- errorStream_ << "RtApiOss::getDeviceInfo: no supported sample rates found for device (" << ainfo.name << ").";
- errorText_ = errorStream_.str();
- error( RtAudioError::WARNING );
- }
- else {
- info.probed = true;
- info.name = ainfo.name;
- }
-
- return info;
-}
-
-
-bool RtApiOss :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,
- unsigned int firstChannel, unsigned int sampleRate,
- RtAudioFormat format, unsigned int *bufferSize,
- RtAudio::StreamOptions *options )
-{
- int mixerfd = open( "/dev/mixer", O_RDWR, 0 );
- if ( mixerfd == -1 ) {
- errorText_ = "RtApiOss::probeDeviceOpen: error opening '/dev/mixer'.";
- return FAILURE;
- }
-
- oss_sysinfo sysinfo;
- int result = ioctl( mixerfd, SNDCTL_SYSINFO, &sysinfo );
- if ( result == -1 ) {
- close( mixerfd );
- errorText_ = "RtApiOss::probeDeviceOpen: error getting sysinfo, OSS version >= 4.0 is required.";
- return FAILURE;
- }
-
- unsigned nDevices = sysinfo.numaudios;
- if ( nDevices == 0 ) {
- // This should not happen because a check is made before this function is called.
- close( mixerfd );
- errorText_ = "RtApiOss::probeDeviceOpen: no devices found!";
- return FAILURE;
- }
-
- if ( device >= nDevices ) {
- // This should not happen because a check is made before this function is called.
- close( mixerfd );
- errorText_ = "RtApiOss::probeDeviceOpen: device ID is invalid!";
- return FAILURE;
- }
-
- oss_audioinfo ainfo;
- ainfo.dev = device;
- result = ioctl( mixerfd, SNDCTL_AUDIOINFO, &ainfo );
- close( mixerfd );
- if ( result == -1 ) {
- errorStream_ << "RtApiOss::getDeviceInfo: error getting device (" << ainfo.name << ") info.";
- errorText_ = errorStream_.str();
- return FAILURE;
- }
-
- // Check if device supports input or output
- if ( ( mode == OUTPUT && !( ainfo.caps & PCM_CAP_OUTPUT ) ) ||
- ( mode == INPUT && !( ainfo.caps & PCM_CAP_INPUT ) ) ) {
- if ( mode == OUTPUT )
- errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") does not support output.";
- else
- errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") does not support input.";
- errorText_ = errorStream_.str();
- return FAILURE;
- }
-
- int flags = 0;
- OssHandle *handle = (OssHandle *) stream_.apiHandle;
- if ( mode == OUTPUT )
- flags |= O_WRONLY;
- else { // mode == INPUT
- if (stream_.mode == OUTPUT && stream_.device[0] == device) {
- // We just set the same device for playback ... close and reopen for duplex (OSS only).
- close( handle->id[0] );
- handle->id[0] = 0;
- if ( !( ainfo.caps & PCM_CAP_DUPLEX ) ) {
- errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") does not support duplex mode.";
- errorText_ = errorStream_.str();
- return FAILURE;
- }
- // Check that the number previously set channels is the same.
- if ( stream_.nUserChannels[0] != channels ) {
- errorStream_ << "RtApiOss::probeDeviceOpen: input/output channels must be equal for OSS duplex device (" << ainfo.name << ").";
- errorText_ = errorStream_.str();
- return FAILURE;
- }
- flags |= O_RDWR;
- }
- else
- flags |= O_RDONLY;
- }
-
- // Set exclusive access if specified.
- if ( options && options->flags & RTAUDIO_HOG_DEVICE ) flags |= O_EXCL;
-
- // Try to open the device.
- int fd;
- fd = open( ainfo.devnode, flags, 0 );
- if ( fd == -1 ) {
- if ( errno == EBUSY )
- errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") is busy.";
- else
- errorStream_ << "RtApiOss::probeDeviceOpen: error opening device (" << ainfo.name << ").";
- errorText_ = errorStream_.str();
- return FAILURE;
- }
-
- // For duplex operation, specifically set this mode (this doesn't seem to work).
- /*
- if ( flags | O_RDWR ) {
- result = ioctl( fd, SNDCTL_DSP_SETDUPLEX, NULL );
- if ( result == -1) {
- errorStream_ << "RtApiOss::probeDeviceOpen: error setting duplex mode for device (" << ainfo.name << ").";
- errorText_ = errorStream_.str();
- return FAILURE;
- }
- }
- */
-
- // Check the device channel support.
- stream_.nUserChannels[mode] = channels;
- if ( ainfo.max_channels < (int)(channels + firstChannel) ) {
- close( fd );
- errorStream_ << "RtApiOss::probeDeviceOpen: the device (" << ainfo.name << ") does not support requested channel parameters.";
- errorText_ = errorStream_.str();
- return FAILURE;
- }
-
- // Set the number of channels.
- int deviceChannels = channels + firstChannel;
- result = ioctl( fd, SNDCTL_DSP_CHANNELS, &deviceChannels );
- if ( result == -1 || deviceChannels < (int)(channels + firstChannel) ) {
- close( fd );
- errorStream_ << "RtApiOss::probeDeviceOpen: error setting channel parameters on device (" << ainfo.name << ").";
- errorText_ = errorStream_.str();
- return FAILURE;
- }
- stream_.nDeviceChannels[mode] = deviceChannels;
-
- // Get the data format mask
- int mask;
- result = ioctl( fd, SNDCTL_DSP_GETFMTS, &mask );
- if ( result == -1 ) {
- close( fd );
- errorStream_ << "RtApiOss::probeDeviceOpen: error getting device (" << ainfo.name << ") data formats.";
- errorText_ = errorStream_.str();
- return FAILURE;
- }
-
- // Determine how to set the device format.
- stream_.userFormat = format;
- int deviceFormat = -1;
- stream_.doByteSwap[mode] = false;
- if ( format == RTAUDIO_SINT8 ) {
- if ( mask & AFMT_S8 ) {
- deviceFormat = AFMT_S8;
- stream_.deviceFormat[mode] = RTAUDIO_SINT8;
- }
- }
- else if ( format == RTAUDIO_SINT16 ) {
- if ( mask & AFMT_S16_NE ) {
- deviceFormat = AFMT_S16_NE;
- stream_.deviceFormat[mode] = RTAUDIO_SINT16;
- }
- else if ( mask & AFMT_S16_OE ) {
- deviceFormat = AFMT_S16_OE;
- stream_.deviceFormat[mode] = RTAUDIO_SINT16;
- stream_.doByteSwap[mode] = true;
- }
- }
- else if ( format == RTAUDIO_SINT24 ) {
- if ( mask & AFMT_S24_NE ) {
- deviceFormat = AFMT_S24_NE;
- stream_.deviceFormat[mode] = RTAUDIO_SINT24;
- }
- else if ( mask & AFMT_S24_OE ) {
- deviceFormat = AFMT_S24_OE;
- stream_.deviceFormat[mode] = RTAUDIO_SINT24;
- stream_.doByteSwap[mode] = true;
- }
- }
- else if ( format == RTAUDIO_SINT32 ) {
- if ( mask & AFMT_S32_NE ) {
- deviceFormat = AFMT_S32_NE;
- stream_.deviceFormat[mode] = RTAUDIO_SINT32;
- }
- else if ( mask & AFMT_S32_OE ) {
- deviceFormat = AFMT_S32_OE;
- stream_.deviceFormat[mode] = RTAUDIO_SINT32;
- stream_.doByteSwap[mode] = true;
- }
- }
-
- if ( deviceFormat == -1 ) {
- // The user requested format is not natively supported by the device.
- if ( mask & AFMT_S16_NE ) {
- deviceFormat = AFMT_S16_NE;
- stream_.deviceFormat[mode] = RTAUDIO_SINT16;
- }
- else if ( mask & AFMT_S32_NE ) {
- deviceFormat = AFMT_S32_NE;
- stream_.deviceFormat[mode] = RTAUDIO_SINT32;
- }
- else if ( mask & AFMT_S24_NE ) {
- deviceFormat = AFMT_S24_NE;
- stream_.deviceFormat[mode] = RTAUDIO_SINT24;
- }
- else if ( mask & AFMT_S16_OE ) {
- deviceFormat = AFMT_S16_OE;
- stream_.deviceFormat[mode] = RTAUDIO_SINT16;
- stream_.doByteSwap[mode] = true;
- }
- else if ( mask & AFMT_S32_OE ) {
- deviceFormat = AFMT_S32_OE;
- stream_.deviceFormat[mode] = RTAUDIO_SINT32;
- stream_.doByteSwap[mode] = true;
- }
- else if ( mask & AFMT_S24_OE ) {
- deviceFormat = AFMT_S24_OE;
- stream_.deviceFormat[mode] = RTAUDIO_SINT24;
- stream_.doByteSwap[mode] = true;
- }
- else if ( mask & AFMT_S8) {
- deviceFormat = AFMT_S8;
- stream_.deviceFormat[mode] = RTAUDIO_SINT8;
- }
- }
-
- if ( stream_.deviceFormat[mode] == 0 ) {
- // This really shouldn't happen ...
- close( fd );
- errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") data format not supported by RtAudio.";
- errorText_ = errorStream_.str();
- return FAILURE;
- }
-
- // Set the data format.
- int temp = deviceFormat;
- result = ioctl( fd, SNDCTL_DSP_SETFMT, &deviceFormat );
- if ( result == -1 || deviceFormat != temp ) {
- close( fd );
- errorStream_ << "RtApiOss::probeDeviceOpen: error setting data format on device (" << ainfo.name << ").";
- errorText_ = errorStream_.str();
- return FAILURE;
- }
-
- // Attempt to set the buffer size. According to OSS, the minimum
- // number of buffers is two. The supposed minimum buffer size is 16
- // bytes, so that will be our lower bound. The argument to this
- // call is in the form 0xMMMMSSSS (hex), where the buffer size (in
- // bytes) is given as 2^SSSS and the number of buffers as 2^MMMM.
- // We'll check the actual value used near the end of the setup
- // procedure.
- int ossBufferBytes = *bufferSize * formatBytes( stream_.deviceFormat[mode] ) * deviceChannels;
- if ( ossBufferBytes < 16 ) ossBufferBytes = 16;
- int buffers = 0;
- if ( options ) buffers = options->numberOfBuffers;
- if ( options && options->flags & RTAUDIO_MINIMIZE_LATENCY ) buffers = 2;
- if ( buffers < 2 ) buffers = 3;
- temp = ((int) buffers << 16) + (int)( log10( (double)ossBufferBytes ) / log10( 2.0 ) );
- result = ioctl( fd, SNDCTL_DSP_SETFRAGMENT, &temp );
- if ( result == -1 ) {
- close( fd );
- errorStream_ << "RtApiOss::probeDeviceOpen: error setting buffer size on device (" << ainfo.name << ").";
- errorText_ = errorStream_.str();
- return FAILURE;
- }
- stream_.nBuffers = buffers;
-
- // Save buffer size (in sample frames).
- *bufferSize = ossBufferBytes / ( formatBytes(stream_.deviceFormat[mode]) * deviceChannels );
- stream_.bufferSize = *bufferSize;
-
- // Set the sample rate.
- int srate = sampleRate;
- result = ioctl( fd, SNDCTL_DSP_SPEED, &srate );
- if ( result == -1 ) {
- close( fd );
- errorStream_ << "RtApiOss::probeDeviceOpen: error setting sample rate (" << sampleRate << ") on device (" << ainfo.name << ").";
- errorText_ = errorStream_.str();
- return FAILURE;
- }
-
- // Verify the sample rate setup worked.
- if ( abs( srate - sampleRate ) > 100 ) {
- close( fd );
- errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") does not support sample rate (" << sampleRate << ").";
- errorText_ = errorStream_.str();
- return FAILURE;
- }
- stream_.sampleRate = sampleRate;
-
- if ( mode == INPUT && stream_.mode == OUTPUT && stream_.device[0] == device) {
- // We're doing duplex setup here.
- stream_.deviceFormat[0] = stream_.deviceFormat[1];
- stream_.nDeviceChannels[0] = deviceChannels;
- }
-
- // Set interleaving parameters.
- stream_.userInterleaved = true;
- stream_.deviceInterleaved[mode] = true;
- if ( options && options->flags & RTAUDIO_NONINTERLEAVED )
- stream_.userInterleaved = false;
-
- // Set flags for buffer conversion
- stream_.doConvertBuffer[mode] = false;
- if ( stream_.userFormat != stream_.deviceFormat[mode] )
- stream_.doConvertBuffer[mode] = true;
- if ( stream_.nUserChannels[mode] < stream_.nDeviceChannels[mode] )
- stream_.doConvertBuffer[mode] = true;
- if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] &&
- stream_.nUserChannels[mode] > 1 )
- stream_.doConvertBuffer[mode] = true;
-
- // Allocate the stream handles if necessary and then save.
- if ( stream_.apiHandle == 0 ) {
- try {
- handle = new OssHandle;
- }
- catch ( std::bad_alloc& ) {
- errorText_ = "RtApiOss::probeDeviceOpen: error allocating OssHandle memory.";
- goto error;
- }
-
- if ( pthread_cond_init( &handle->runnable, NULL ) ) {
- errorText_ = "RtApiOss::probeDeviceOpen: error initializing pthread condition variable.";
- goto error;
- }
-
- stream_.apiHandle = (void *) handle;
- }
- else {
- handle = (OssHandle *) stream_.apiHandle;
- }
- handle->id[mode] = fd;
-
- // Allocate necessary internal buffers.
- unsigned long bufferBytes;
- bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat );
- stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 );
- if ( stream_.userBuffer[mode] == NULL ) {
- errorText_ = "RtApiOss::probeDeviceOpen: error allocating user buffer memory.";
- goto error;
- }
-
- if ( stream_.doConvertBuffer[mode] ) {
-
- bool makeBuffer = true;
- bufferBytes = stream_.nDeviceChannels[mode] * formatBytes( stream_.deviceFormat[mode] );
- if ( mode == INPUT ) {
- if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) {
- unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] );
- if ( bufferBytes <= bytesOut ) makeBuffer = false;
- }
- }
-
- if ( makeBuffer ) {
- bufferBytes *= *bufferSize;
- if ( stream_.deviceBuffer ) free( stream_.deviceBuffer );
- stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 );
- if ( stream_.deviceBuffer == NULL ) {
- errorText_ = "RtApiOss::probeDeviceOpen: error allocating device buffer memory.";
- goto error;
- }
- }
- }
-
- stream_.device[mode] = device;
- stream_.state = STREAM_STOPPED;
-
- // Setup the buffer conversion information structure.
- if ( stream_.doConvertBuffer[mode] ) setConvertInfo( mode, firstChannel );
-
- // Setup thread if necessary.
- if ( stream_.mode == OUTPUT && mode == INPUT ) {
- // We had already set up an output stream.
- stream_.mode = DUPLEX;
- if ( stream_.device[0] == device ) handle->id[0] = fd;
- }
- else {
- stream_.mode = mode;
-
- // Setup callback thread.
- stream_.callbackInfo.object = (void *) this;
-
- // Set the thread attributes for joinable and realtime scheduling
- // priority. The higher priority will only take affect if the
- // program is run as root or suid.
- pthread_attr_t attr;
- pthread_attr_init( &attr );
- pthread_attr_setdetachstate( &attr, PTHREAD_CREATE_JOINABLE );
-#ifdef SCHED_RR // Undefined with some OSes (eg: NetBSD 1.6.x with GNU Pthread)
- if ( options && options->flags & RTAUDIO_SCHEDULE_REALTIME ) {
- struct sched_param param;
- int priority = options->priority;
- int min = sched_get_priority_min( SCHED_RR );
- int max = sched_get_priority_max( SCHED_RR );
- if ( priority < min ) priority = min;
- else if ( priority > max ) priority = max;
- param.sched_priority = priority;
- pthread_attr_setschedparam( &attr, &param );
- 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;
- }
- }
-}
-
- // Indentation settings for Vim and Emacs
- //
- // Local Variables:
- // c-basic-offset: 2
- // indent-tabs-mode: nil
- // End:
- //
- // vim: et sts=2 sw=2
-
+/************************************************************************/
+/*! \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 GitHub site: https://github.com/thestk/rtaudio
+ RtAudio WWW site: http://www.music.mcgill.ca/~gary/rtaudio/
+
+ RtAudio: realtime audio i/o C++ classes
+ Copyright (c) 2001-2023 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 6.0.1
+
+#include "RtAudio.h"
+#include <iostream>
+#include <cstdlib>
+#include <cstring>
+#include <climits>
+#include <cmath>
+#include <algorithm>
+#include <codecvt>
+#include <locale>
+
+#if defined(_WIN32)
+#include <windows.h>
+#endif
+
+// 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
+};
+
+template<typename T> inline
+std::string convertCharPointerToStdString(const T *text);
+
+template<> inline
+std::string convertCharPointerToStdString(const char *text)
+{
+ return text;
+}
+
+template<> inline
+std::string convertCharPointerToStdString(const wchar_t *text)
+{
+ return std::wstring_convert<std::codecvt_utf8_utf16<wchar_t>>{}.to_bytes(text);
+}
+
+#if defined(_MSC_VER)
+ #define MUTEX_INITIALIZE(A) InitializeCriticalSection(A)
+ #define MUTEX_DESTROY(A) DeleteCriticalSection(A)
+ #define MUTEX_LOCK(A) EnterCriticalSection(A)
+ #define MUTEX_UNLOCK(A) LeaveCriticalSection(A)
+#else
+ #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)
+#endif
+
+// *************************************************** //
+//
+// RtApi subclass prototypes.
+//
+// *************************************************** //
+
+#if defined(__MACOSX_CORE__)
+
+#include <CoreAudio/AudioHardware.h>
+
+class RtApiCore: public RtApi
+{
+public:
+
+ RtApiCore();
+ ~RtApiCore();
+ RtAudio::Api getCurrentApi( void ) override { return RtAudio::MACOSX_CORE; }
+ unsigned int getDefaultOutputDevice( void ) override;
+ unsigned int getDefaultInputDevice( void ) override;
+ void closeStream( void ) override;
+ RtAudioErrorType startStream( void ) override;
+ RtAudioErrorType stopStream( void ) override;
+ RtAudioErrorType abortStream( void ) override;
+
+ // This function is intended for internal use only. It must be
+ // public because it is called by an internal callback handler,
+ // which is not a member of RtAudio. External use of this function
+ // will most likely produce highly undesirable results!
+ bool callbackEvent( AudioDeviceID deviceId,
+ const AudioBufferList *inBufferList,
+ const AudioBufferList *outBufferList );
+
+ private:
+ void probeDevices( void ) override;
+ bool probeDeviceInfo( AudioDeviceID id, RtAudio::DeviceInfo &info );
+ bool probeDeviceOpen( unsigned int deviceId, StreamMode mode, unsigned int channels,
+ unsigned int firstChannel, unsigned int sampleRate,
+ RtAudioFormat format, unsigned int *bufferSize,
+ RtAudio::StreamOptions *options ) override;
+ static const char* getErrorCode( OSStatus code );
+ std::vector< AudioDeviceID > deviceIds_;
+};
+
+#endif
+
+#if defined(__UNIX_JACK__)
+
+#include <jack/jack.h>
+
+class RtApiJack: public RtApi
+{
+public:
+
+ RtApiJack();
+ ~RtApiJack();
+ RtAudio::Api getCurrentApi( void ) override { return RtAudio::UNIX_JACK; }
+ void closeStream( void ) override;
+ RtAudioErrorType startStream( void ) override;
+ RtAudioErrorType stopStream( void ) override;
+ RtAudioErrorType abortStream( void ) override;
+
+ // This function is intended for internal use only. It must be
+ // public because it is called by the internal callback handler,
+ // which is not a member of RtAudio. External use of this function
+ // will most likely produce highly undesirable results!
+ bool callbackEvent( unsigned long nframes );
+
+ private:
+ void probeDevices( void ) override;
+ bool probeDeviceInfo( RtAudio::DeviceInfo &info, jack_client_t *client );
+ bool probeDeviceOpen( unsigned int deviceId, StreamMode mode, unsigned int channels,
+ unsigned int firstChannel, unsigned int sampleRate,
+ RtAudioFormat format, unsigned int *bufferSize,
+ RtAudio::StreamOptions *options ) override;
+
+ bool shouldAutoconnect_;
+};
+
+#endif
+
+#if defined(__WINDOWS_ASIO__)
+
+class RtApiAsio: public RtApi
+{
+public:
+
+ RtApiAsio();
+ ~RtApiAsio();
+ RtAudio::Api getCurrentApi( void ) override { return RtAudio::WINDOWS_ASIO; }
+ void closeStream( void ) override;
+ RtAudioErrorType startStream( void ) override;
+ RtAudioErrorType stopStream( void ) override;
+ RtAudioErrorType abortStream( void ) override;
+
+ // This function is intended for internal use only. It must be
+ // public because it is called by the internal callback handler,
+ // which is not a member of RtAudio. External use of this function
+ // will most likely produce highly undesirable results!
+ bool callbackEvent( long bufferIndex );
+
+ private:
+
+ bool coInitialized_;
+ void probeDevices( void ) override;
+ bool probeDeviceInfo( RtAudio::DeviceInfo &info );
+ bool probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,
+ unsigned int firstChannel, unsigned int sampleRate,
+ RtAudioFormat format, unsigned int *bufferSize,
+ RtAudio::StreamOptions *options ) override;
+};
+
+#endif
+
+#if defined(__WINDOWS_DS__)
+
+class RtApiDs: public RtApi
+{
+public:
+
+ RtApiDs();
+ ~RtApiDs();
+ RtAudio::Api getCurrentApi( void ) override { return RtAudio::WINDOWS_DS; }
+ void closeStream( void ) override;
+ RtAudioErrorType startStream( void ) override;
+ RtAudioErrorType stopStream( void ) override;
+ RtAudioErrorType abortStream( void ) override;
+
+ // This function is intended for internal use only. It must be
+ // public because it is called by the internal callback handler,
+ // which is not a member of RtAudio. External use of this function
+ // will most likely produce highly undesirable results!
+ void callbackEvent( void );
+
+ private:
+
+ bool coInitialized_;
+ bool buffersRolling;
+ long duplexPrerollBytes;
+ std::vector<struct DsDevice> dsDevices_;
+
+ void probeDevices( void ) override;
+ bool probeDeviceInfo( RtAudio::DeviceInfo &info, DsDevice &dsDevice );
+ bool probeDeviceOpen( unsigned int deviceId, StreamMode mode, unsigned int channels,
+ unsigned int firstChannel, unsigned int sampleRate,
+ RtAudioFormat format, unsigned int *bufferSize,
+ RtAudio::StreamOptions *options ) override;
+};
+
+#endif
+
+#if defined(__WINDOWS_WASAPI__)
+
+struct IMMDeviceEnumerator;
+
+class RtApiWasapi : public RtApi
+{
+public:
+ RtApiWasapi();
+ virtual ~RtApiWasapi();
+ RtAudio::Api getCurrentApi( void ) override { return RtAudio::WINDOWS_WASAPI; }
+ unsigned int getDefaultOutputDevice( void ) override;
+ unsigned int getDefaultInputDevice( void ) override;
+ void closeStream( void ) override;
+ RtAudioErrorType startStream( void ) override;
+ RtAudioErrorType stopStream( void ) override;
+ RtAudioErrorType abortStream( void ) override;
+
+private:
+ bool coInitialized_;
+ IMMDeviceEnumerator* deviceEnumerator_;
+ std::vector< std::pair< std::string, bool> > deviceIds_;
+
+ void probeDevices( void ) override;
+ bool probeDeviceInfo( RtAudio::DeviceInfo &info, LPWSTR deviceId, bool isCaptureDevice );
+ bool probeDeviceOpen( unsigned int deviceId, StreamMode mode, unsigned int channels,
+ unsigned int firstChannel, unsigned int sampleRate,
+ RtAudioFormat format, unsigned int* bufferSize,
+ RtAudio::StreamOptions* options ) override;
+
+ static DWORD WINAPI runWasapiThread( void* wasapiPtr );
+ static DWORD WINAPI stopWasapiThread( void* wasapiPtr );
+ static DWORD WINAPI abortWasapiThread( void* wasapiPtr );
+ void wasapiThread();
+};
+
+#endif
+
+#if defined(__LINUX_ALSA__)
+
+class RtApiAlsa: public RtApi
+{
+public:
+
+ RtApiAlsa();
+ ~RtApiAlsa();
+ RtAudio::Api getCurrentApi() override { return RtAudio::LINUX_ALSA; }
+ void closeStream( void ) override;
+ RtAudioErrorType startStream( void ) override;
+ RtAudioErrorType stopStream( void ) override;
+ RtAudioErrorType abortStream( void ) override;
+
+ // This function is intended for internal use only. It must be
+ // public because it is called by the internal callback handler,
+ // which is not a member of RtAudio. External use of this function
+ // will most likely produce highly undesirable results!
+ void callbackEvent( void );
+
+ private:
+ std::vector<std::pair<std::string, unsigned int>> deviceIdPairs_;
+
+ void probeDevices( void ) override;
+ bool probeDeviceInfo( RtAudio::DeviceInfo &info, std::string name );
+ bool probeDeviceOpen( unsigned int deviceId, StreamMode mode, unsigned int channels,
+ unsigned int firstChannel, unsigned int sampleRate,
+ RtAudioFormat format, unsigned int *bufferSize,
+ RtAudio::StreamOptions *options ) override;
+};
+
+#endif
+
+#if defined(__LINUX_PULSE__)
+
+#include <pulse/pulseaudio.h>
+
+class RtApiPulse: public RtApi
+{
+public:
+ ~RtApiPulse();
+ RtAudio::Api getCurrentApi() override { return RtAudio::LINUX_PULSE; }
+ void closeStream( void ) override;
+ RtAudioErrorType startStream( void ) override;
+ RtAudioErrorType stopStream( void ) override;
+ RtAudioErrorType abortStream( void ) override;
+
+ // This function is intended for internal use only. It must be
+ // public because it is called by the internal callback handler,
+ // which is not a member of RtAudio. External use of this function
+ // will most likely produce highly undesirable results!
+ void callbackEvent( void );
+
+ struct PaDeviceInfo {
+ std::string sinkName;
+ std::string sourceName;
+ };
+
+ private:
+ std::vector< PaDeviceInfo > paDeviceList_;
+
+ void probeDevices( void ) override;
+ bool probeDeviceOpen( unsigned int deviceId, StreamMode mode, unsigned int channels,
+ unsigned int firstChannel, unsigned int sampleRate,
+ RtAudioFormat format, unsigned int *bufferSize,
+ RtAudio::StreamOptions *options ) override;
+};
+
+#endif
+
+#if defined(__LINUX_OSS__)
+
+#include <sys/soundcard.h>
+
+class RtApiOss: public RtApi
+{
+public:
+
+ RtApiOss();
+ ~RtApiOss();
+ RtAudio::Api getCurrentApi() override { return RtAudio::LINUX_OSS; }
+ void closeStream( void ) override;
+ RtAudioErrorType startStream( void ) override;
+ RtAudioErrorType stopStream( void ) override;
+ RtAudioErrorType abortStream( void ) override;
+
+ // This function is intended for internal use only. It must be
+ // public because it is called by the internal callback handler,
+ // which is not a member of RtAudio. External use of this function
+ // will most likely produce highly undesirable results!
+ void callbackEvent( void );
+
+ private:
+
+ void probeDevices( void ) override;
+ bool probeDeviceInfo( RtAudio::DeviceInfo &info, oss_audioinfo &ainfo );
+ bool probeDeviceOpen( unsigned int deviceId, StreamMode mode, unsigned int channels,
+ unsigned int firstChannel, unsigned int sampleRate,
+ RtAudioFormat format, unsigned int *bufferSize,
+ RtAudio::StreamOptions *options ) override;
+};
+
+#endif
+
+#if defined(__RTAUDIO_DUMMY__)
+
+class RtApiDummy: public RtApi
+{
+public:
+
+ RtApiDummy() { errorText_ = "RtApiDummy: This class provides no functionality."; error( RTAUDIO_WARNING ); }
+ RtAudio::Api getCurrentApi( void ) override { return RtAudio::RTAUDIO_DUMMY; }
+ void closeStream( void ) override {}
+ RtAudioErrorType startStream( void ) override { return RTAUDIO_NO_ERROR; }
+ RtAudioErrorType stopStream( void ) override { return RTAUDIO_NO_ERROR; }
+ RtAudioErrorType abortStream( void ) override { return RTAUDIO_NO_ERROR; }
+
+ private:
+
+ bool probeDeviceOpen( unsigned int /*deviceId*/, StreamMode /*mode*/, unsigned int /*channels*/,
+ unsigned int /*firstChannel*/, unsigned int /*sampleRate*/,
+ RtAudioFormat /*format*/, unsigned int * /*bufferSize*/,
+ RtAudio::StreamOptions * /*options*/ ) override { return false; }
+};
+
+#endif
+
+// *************************************************** //
+//
+// RtAudio definitions.
+//
+// *************************************************** //
+
+std::string RtAudio :: getVersion( void )
+{
+ return RTAUDIO_VERSION;
+}
+
+// Define API names and display names.
+// Must be in same order as API enum.
+extern "C" {
+const char* rtaudio_api_names[][2] = {
+ { "unspecified" , "Unknown" },
+ { "core" , "CoreAudio" },
+ { "alsa" , "ALSA" },
+ { "jack" , "Jack" },
+ { "pulse" , "Pulse" },
+ { "oss" , "OpenSoundSystem" },
+ { "asio" , "ASIO" },
+ { "wasapi" , "WASAPI" },
+ { "ds" , "DirectSound" },
+ { "dummy" , "Dummy" },
+};
+
+const unsigned int rtaudio_num_api_names =
+ sizeof(rtaudio_api_names)/sizeof(rtaudio_api_names[0]);
+
+// The order here will control the order of RtAudio's API search in
+// the constructor.
+extern "C" const RtAudio::Api rtaudio_compiled_apis[] = {
+#if defined(__MACOSX_CORE__)
+ RtAudio::MACOSX_CORE,
+#endif
+#if defined(__LINUX_ALSA__)
+ RtAudio::LINUX_ALSA,
+#endif
+#if defined(__UNIX_JACK__)
+ RtAudio::UNIX_JACK,
+#endif
+#if defined(__LINUX_PULSE__)
+ RtAudio::LINUX_PULSE,
+#endif
+#if defined(__LINUX_OSS__)
+ RtAudio::LINUX_OSS,
+#endif
+#if defined(__WINDOWS_ASIO__)
+ RtAudio::WINDOWS_ASIO,
+#endif
+#if defined(__WINDOWS_WASAPI__)
+ RtAudio::WINDOWS_WASAPI,
+#endif
+#if defined(__WINDOWS_DS__)
+ RtAudio::WINDOWS_DS,
+#endif
+#if defined(__RTAUDIO_DUMMY__)
+ RtAudio::RTAUDIO_DUMMY,
+#endif
+ RtAudio::UNSPECIFIED,
+};
+
+extern "C" const unsigned int rtaudio_num_compiled_apis =
+ sizeof(rtaudio_compiled_apis)/sizeof(rtaudio_compiled_apis[0])-1;
+}
+
+// This is a compile-time check that rtaudio_num_api_names == RtAudio::NUM_APIS.
+// If the build breaks here, check that they match.
+template<bool b> class StaticAssert { private: StaticAssert() {} };
+template<> class StaticAssert<true>{ public: StaticAssert() {} };
+class StaticAssertions { StaticAssertions() {
+ StaticAssert<rtaudio_num_api_names == RtAudio::NUM_APIS>();
+}};
+
+void RtAudio :: getCompiledApi( std::vector<RtAudio::Api> &apis )
+{
+ apis = std::vector<RtAudio::Api>(rtaudio_compiled_apis,
+ rtaudio_compiled_apis + rtaudio_num_compiled_apis);
+}
+
+std::string RtAudio :: getApiName( RtAudio::Api api )
+{
+ if (api < 0 || api >= RtAudio::NUM_APIS)
+ return "";
+ return rtaudio_api_names[api][0];
+}
+
+std::string RtAudio :: getApiDisplayName( RtAudio::Api api )
+{
+ if (api < 0 || api >= RtAudio::NUM_APIS)
+ return "Unknown";
+ return rtaudio_api_names[api][1];
+}
+
+RtAudio::Api RtAudio :: getCompiledApiByName( const std::string &name )
+{
+ unsigned int i=0;
+ for (i = 0; i < rtaudio_num_compiled_apis; ++i)
+ if (name == rtaudio_api_names[rtaudio_compiled_apis[i]][0])
+ return rtaudio_compiled_apis[i];
+ return RtAudio::UNSPECIFIED;
+}
+
+RtAudio::Api RtAudio :: getCompiledApiByDisplayName( const std::string &name )
+{
+ unsigned int i=0;
+ for (i = 0; i < rtaudio_num_compiled_apis; ++i)
+ if (name == rtaudio_api_names[rtaudio_compiled_apis[i]][1])
+ return rtaudio_compiled_apis[i];
+ return RtAudio::UNSPECIFIED;
+}
+
+void RtAudio :: openRtApi( RtAudio::Api api )
+{
+ if ( rtapi_ )
+ delete rtapi_;
+ rtapi_ = 0;
+
+#if defined(__UNIX_JACK__)
+ if ( api == UNIX_JACK )
+ rtapi_ = new RtApiJack();
+#endif
+#if defined(__LINUX_ALSA__)
+ if ( api == LINUX_ALSA )
+ rtapi_ = new RtApiAlsa();
+#endif
+#if defined(__LINUX_PULSE__)
+ if ( api == LINUX_PULSE )
+ rtapi_ = new RtApiPulse();
+#endif
+#if defined(__LINUX_OSS__)
+ if ( api == LINUX_OSS )
+ rtapi_ = new RtApiOss();
+#endif
+#if defined(__WINDOWS_ASIO__)
+ if ( api == WINDOWS_ASIO )
+ rtapi_ = new RtApiAsio();
+#endif
+#if defined(__WINDOWS_WASAPI__)
+ if ( api == WINDOWS_WASAPI )
+ rtapi_ = new RtApiWasapi();
+#endif
+#if defined(__WINDOWS_DS__)
+ if ( api == WINDOWS_DS )
+ rtapi_ = new RtApiDs();
+#endif
+#if defined(__MACOSX_CORE__)
+ if ( api == MACOSX_CORE )
+ rtapi_ = new RtApiCore();
+#endif
+#if defined(__RTAUDIO_DUMMY__)
+ if ( api == RTAUDIO_DUMMY )
+ rtapi_ = new RtApiDummy();
+#endif
+}
+
+RtAudio :: RtAudio( RtAudio::Api api, RtAudioErrorCallback&& errorCallback )
+{
+ rtapi_ = 0;
+
+ std::string errorMessage;
+ if ( api != UNSPECIFIED ) {
+ // Attempt to open the specified API.
+ openRtApi( api );
+
+ if ( rtapi_ ) {
+ if ( errorCallback ) rtapi_->setErrorCallback( errorCallback );
+ return;
+ }
+
+ // No compiled support for specified API value. Issue a warning
+ // and continue as if no API was specified.
+ errorMessage = "RtAudio: no compiled support for specified API argument!";
+ if ( errorCallback )
+ errorCallback( RTAUDIO_INVALID_USE, errorMessage );
+ else
+ std::cerr << '\n' << errorMessage << '\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_->getDeviceNames()).size() > 0 )
+ break;
+ }
+
+ if ( rtapi_ ) {
+ if ( errorCallback ) rtapi_->setErrorCallback( errorCallback );
+ return;
+ }
+
+ // It should not be possible to get here because the preprocessor
+ // definition __RTAUDIO_DUMMY__ is automatically defined in RtAudio.h
+ // if no API-specific definitions are passed to the compiler. But just
+ // in case something weird happens, issue an error message and abort.
+ errorMessage = "RtAudio: no compiled API support found ... critical error!";
+ if ( errorCallback )
+ errorCallback( RTAUDIO_INVALID_USE, errorMessage );
+ else
+ std::cerr << '\n' << errorMessage << '\n' << std::endl;
+ abort();
+}
+
+RtAudio :: ~RtAudio()
+{
+ if ( rtapi_ )
+ delete rtapi_;
+}
+
+RtAudioErrorType RtAudio :: openStream( RtAudio::StreamParameters *outputParameters,
+ RtAudio::StreamParameters *inputParameters,
+ RtAudioFormat format, unsigned int sampleRate,
+ unsigned int *bufferFrames,
+ RtAudioCallback callback, void *userData,
+ RtAudio::StreamOptions *options )
+{
+ return rtapi_->openStream( outputParameters, inputParameters, format,
+ sampleRate, bufferFrames, callback,
+ userData, options );
+}
+
+// *************************************************** //
+//
+// Public RtApi definitions (see end of file for
+// private or protected utility functions).
+//
+// *************************************************** //
+
+RtApi :: RtApi()
+{
+ clearStreamInfo();
+ MUTEX_INITIALIZE( &stream_.mutex );
+ errorCallback_ = 0;
+ showWarnings_ = true;
+ currentDeviceId_ = 129;
+}
+
+RtApi :: ~RtApi()
+{
+ MUTEX_DESTROY( &stream_.mutex );
+}
+
+RtAudioErrorType RtApi :: openStream( RtAudio::StreamParameters *oParams,
+ RtAudio::StreamParameters *iParams,
+ RtAudioFormat format, unsigned int sampleRate,
+ unsigned int *bufferFrames,
+ RtAudioCallback callback, void *userData,
+ RtAudio::StreamOptions *options )
+{
+ if ( stream_.state != STREAM_CLOSED ) {
+ errorText_ = "RtApi::openStream: a stream is already open!";
+ return error( RTAUDIO_INVALID_USE );
+ }
+
+ // 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.";
+ return error( RTAUDIO_INVALID_PARAMETER );
+ }
+
+ if ( iParams && iParams->nChannels < 1 ) {
+ errorText_ = "RtApi::openStream: a non-NULL input StreamParameters structure cannot have an nChannels value less than one.";
+ return error( RTAUDIO_INVALID_PARAMETER );
+ }
+
+ if ( oParams == NULL && iParams == NULL ) {
+ errorText_ = "RtApi::openStream: input and output StreamParameters structures are both NULL!";
+ return error( RTAUDIO_INVALID_PARAMETER );
+ }
+
+ if ( formatBytes(format) == 0 ) {
+ errorText_ = "RtApi::openStream: 'format' parameter value is undefined.";
+ return error( RTAUDIO_INVALID_PARAMETER );
+ }
+
+ // Scan devices if none currently listed.
+ if ( deviceList_.size() == 0 ) probeDevices();
+
+ unsigned int m, oChannels = 0;
+ if ( oParams ) {
+ oChannels = oParams->nChannels;
+ // Verify that the oParams->deviceId is found in our list
+ for ( m=0; m<deviceList_.size(); m++ ) {
+ if ( deviceList_[m].ID == oParams->deviceId ) break;
+ }
+ if ( m == deviceList_.size() ) {
+ errorText_ = "RtApi::openStream: output device ID is invalid.";
+ return error( RTAUDIO_INVALID_PARAMETER );
+ }
+ }
+
+ unsigned int iChannels = 0;
+ if ( iParams ) {
+ iChannels = iParams->nChannels;
+ for ( m=0; m<deviceList_.size(); m++ ) {
+ if ( deviceList_[m].ID == iParams->deviceId ) break;
+ }
+ if ( m == deviceList_.size() ) {
+ errorText_ = "RtApi::openStream: input device ID is invalid.";
+ return error( RTAUDIO_INVALID_PARAMETER );
+ }
+ }
+
+ bool result;
+
+ if ( oChannels > 0 ) {
+
+ result = probeDeviceOpen( oParams->deviceId, OUTPUT, oChannels, oParams->firstChannel,
+ sampleRate, format, bufferFrames, options );
+ if ( result == false )
+ return error( RTAUDIO_SYSTEM_ERROR );
+ }
+
+ if ( iChannels > 0 ) {
+
+ result = probeDeviceOpen( iParams->deviceId, INPUT, iChannels, iParams->firstChannel,
+ sampleRate, format, bufferFrames, options );
+ if ( result == false )
+ return error( RTAUDIO_SYSTEM_ERROR );
+ }
+
+ stream_.callbackInfo.callback = (void *) callback;
+ stream_.callbackInfo.userData = userData;
+
+ if ( options ) options->numberOfBuffers = stream_.nBuffers;
+ stream_.state = STREAM_STOPPED;
+ return RTAUDIO_NO_ERROR;
+}
+
+void RtApi :: probeDevices( void )
+{
+ // This function MUST be implemented in all subclasses! Within each
+ // API, this function will be used to:
+ // - enumerate the devices and fill or update our
+ // std::vector< RtAudio::DeviceInfo> deviceList_ class variable
+ // - store corresponding (usually API-specific) identifiers that
+ // are needed to open each device
+ // - make sure that the default devices are properly identified
+ // within the deviceList_ (unless API-specific functions are
+ // available for this purpose).
+ //
+ // The function should not reprobe devices that have already been
+ // found. The function must properly handle devices that are removed
+ // or added.
+ //
+ // Ideally, we would also configure callback functions to be invoked
+ // when devices are added or removed (which could be used to inform
+ // clients about changes). However, none of the APIs currently
+ // support notification of _new_ devices and I don't see the
+ // usefulness of having this work only for device removal.
+ return;
+}
+
+unsigned int RtApi :: getDeviceCount( void )
+{
+ probeDevices();
+ return (unsigned int)deviceList_.size();
+}
+
+std::vector<unsigned int> RtApi :: getDeviceIds( void )
+{
+ probeDevices();
+
+ // Copy device IDs into output vector.
+ std::vector<unsigned int> deviceIds;
+ for ( unsigned int m=0; m<deviceList_.size(); m++ )
+ deviceIds.push_back( deviceList_[m].ID );
+
+ return deviceIds;
+}
+
+std::vector<std::string> RtApi :: getDeviceNames( void )
+{
+ probeDevices();
+
+ // Copy device names into output vector.
+ std::vector<std::string> deviceNames;
+ for ( unsigned int m=0; m<deviceList_.size(); m++ )
+ deviceNames.push_back( deviceList_[m].name );
+
+ return deviceNames;
+}
+
+unsigned int RtApi :: getDefaultInputDevice( void )
+{
+ // Should be reimplemented in subclasses if necessary.
+ if ( deviceList_.size() == 0 ) probeDevices();
+ for ( unsigned int i = 0; i < deviceList_.size(); i++ ) {
+ if ( deviceList_[i].isDefaultInput )
+ return deviceList_[i].ID;
+ }
+
+ // If not found, find the first device with input channels, set it
+ // as the default, and return the ID.
+ for ( unsigned int i = 0; i < deviceList_.size(); i++ ) {
+ if ( deviceList_[i].inputChannels > 0 ) {
+ deviceList_[i].isDefaultInput = true;
+ return deviceList_[i].ID;
+ }
+ }
+
+ return 0;
+}
+
+unsigned int RtApi :: getDefaultOutputDevice( void )
+{
+ // Should be reimplemented in subclasses if necessary.
+ if ( deviceList_.size() == 0 ) probeDevices();
+ for ( unsigned int i = 0; i < deviceList_.size(); i++ ) {
+ if ( deviceList_[i].isDefaultOutput )
+ return deviceList_[i].ID;
+ }
+
+ // If not found, find the first device with output channels, set it
+ // as the default, and return the ID.
+ for ( unsigned int i = 0; i < deviceList_.size(); i++ ) {
+ if ( deviceList_[i].outputChannels > 0 ) {
+ deviceList_[i].isDefaultOutput = true;
+ return deviceList_[i].ID;
+ }
+ }
+
+ return 0;
+}
+
+RtAudio::DeviceInfo RtApi :: getDeviceInfo( unsigned int deviceId )
+{
+ if ( deviceList_.size() == 0 ) probeDevices();
+ for ( unsigned int m=0; m<deviceList_.size(); m++ ) {
+ if ( deviceList_[m].ID == deviceId )
+ return deviceList_[m];
+ }
+
+ errorText_ = "RtApi::getDeviceInfo: deviceId argument not found.";
+ error( RTAUDIO_INVALID_PARAMETER );
+ return RtAudio::DeviceInfo();
+}
+
+void RtApi :: closeStream( void )
+{
+ // MUST be implemented in subclasses!
+ return;
+}
+
+bool RtApi :: probeDeviceOpen( unsigned int /*deviceId*/, 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 )
+{
+ 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 )
+{
+#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 )
+{
+ if ( time >= 0.0 )
+ stream_.streamTime = time;
+ /*
+#if defined( HAVE_GETTIMEOFDAY )
+ gettimeofday( &stream_.lastTickTimestamp, NULL );
+#endif
+ */
+}
+
+unsigned int RtApi :: getStreamSampleRate( void )
+{
+ if ( isStreamOpen() ) return stream_.sampleRate;
+ else return 0;
+}
+
+
+// *************************************************** //
+//
+// OS/API-specific methods.
+//
+// *************************************************** //
+
+#if defined(__MACOSX_CORE__)
+
+#include <unistd.h>
+
+// 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.
+ bool xrunListenerAdded[2];
+ bool disconnectListenerAdded[2];
+
+ CoreHandle()
+ :deviceBuffer(0), drainCounter(0), internalDrain(false) { nStreams[0] = 1; nStreams[1] = 1; id[0] = 0; id[1] = 0; procId[0] = 0; procId[1] = 0; xrun[0] = false; xrun[1] = false; xrunListenerAdded[0] = false; xrunListenerAdded[1] = false; disconnectListenerAdded[0] = false; disconnectListenerAdded[1] = false; }
+};
+
+#if defined( MAC_OS_VERSION_12_0 ) && ( MAC_OS_X_VERSION_MIN_REQUIRED >= MAC_OS_VERSION_12_0 )
+ #define KAUDIOOBJECTPROPERTYELEMENT kAudioObjectPropertyElementMain
+#else
+ #define KAUDIOOBJECTPROPERTYELEMENT kAudioObjectPropertyElementMaster // deprecated with macOS 12
+#endif
+
+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,
+ KAUDIOOBJECTPROPERTYELEMENT };
+ OSStatus result = AudioObjectSetPropertyData( kAudioObjectSystemObject, &property, 0, NULL, sizeof(CFRunLoopRef), &theRunLoop);
+ if ( result != noErr ) {
+ errorText_ = "RtApiCore::RtApiCore: error setting run loop property!";
+ error( RTAUDIO_SYSTEM_ERROR );
+ }
+#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 :: getDefaultOutputDevice( void )
+{
+ AudioDeviceID id;
+ UInt32 dataSize = sizeof( AudioDeviceID );
+ AudioObjectPropertyAddress property = { kAudioHardwarePropertyDefaultOutputDevice, kAudioObjectPropertyScopeGlobal, KAUDIOOBJECTPROPERTYELEMENT };
+ OSStatus result = AudioObjectGetPropertyData( kAudioObjectSystemObject, &property, 0, NULL, &dataSize, &id );
+ if ( result != noErr ) {
+ errorText_ = "RtApiCore::getDefaultOutputDevice: OS-X system error getting device.";
+ error( RTAUDIO_SYSTEM_ERROR );
+ return 0;
+ }
+
+ for ( unsigned int m=0; m<deviceIds_.size(); m++ ) {
+ if ( deviceIds_[m] == id ) {
+ if ( deviceList_[m].isDefaultOutput == false ) {
+ deviceList_[m].isDefaultOutput = true;
+ for ( unsigned int j=m+1; j<deviceIds_.size(); j++ ) {
+ // make sure any remaining devices are not listed as the default
+ deviceList_[j].isDefaultOutput = false;
+ }
+ }
+ return deviceList_[m].ID;
+ }
+ deviceList_[m].isDefaultOutput = false;
+ }
+
+ // If not found above, then do system probe of devices and try again.
+ probeDevices();
+ for ( unsigned int m=0; m<deviceIds_.size(); m++ ) {
+ if ( deviceIds_[m] == id ) return deviceList_[m].ID;
+ }
+ return 0;
+}
+
+unsigned int RtApiCore :: getDefaultInputDevice( void )
+{
+ AudioDeviceID id;
+ UInt32 dataSize = sizeof( AudioDeviceID );
+ AudioObjectPropertyAddress property = { kAudioHardwarePropertyDefaultInputDevice, kAudioObjectPropertyScopeGlobal, KAUDIOOBJECTPROPERTYELEMENT };
+ OSStatus result = AudioObjectGetPropertyData( kAudioObjectSystemObject, &property, 0, NULL, &dataSize, &id );
+ if ( result != noErr ) {
+ errorText_ = "RtApiCore::getDefaultInputDevice: OS-X system error getting device.";
+ error( RTAUDIO_SYSTEM_ERROR );
+ return 0;
+ }
+
+ for ( unsigned int m=0; m<deviceIds_.size(); m++ ) {
+ if ( deviceIds_[m] == id ) {
+ if ( deviceList_[m].isDefaultInput == false ) {
+ deviceList_[m].isDefaultInput = true;
+ for ( unsigned int j=m+1; j<deviceIds_.size(); j++ ) {
+ // make sure any remaining devices are not listed as the default
+ deviceList_[j].isDefaultInput = false;
+ }
+ }
+ return deviceList_[m].ID;
+ }
+ deviceList_[m].isDefaultInput = false;
+ }
+
+ // If not found above, then do system probe of devices and try again.
+ probeDevices();
+ for ( unsigned int m=0; m<deviceIds_.size(); m++ ) {
+ if ( deviceIds_[m] == id ) return deviceList_[m].ID;
+ }
+ return 0;
+}
+
+// If a device used in an open stream is disconnected, close the stream.
+static OSStatus streamDisconnectListener( AudioObjectID /*id*/,
+ UInt32 nAddresses,
+ const AudioObjectPropertyAddress properties[],
+ void* infoPointer )
+{
+ for ( UInt32 i=0; i<nAddresses; i++ ) {
+ if ( properties[i].mSelector == kAudioDevicePropertyDeviceIsAlive ) {
+ CallbackInfo *info = (CallbackInfo *) infoPointer;
+ RtApiCore *object = (RtApiCore *) info->object;
+ info->deviceDisconnected = true;
+ object->closeStream();
+ return kAudioHardwareUnspecifiedError;
+ }
+ }
+
+ return kAudioHardwareNoError;
+}
+
+void RtApiCore :: probeDevices( void )
+{
+ // See list of required functionality in RtApi::probeDevices().
+
+ // Find out how many audio devices there are.
+ UInt32 dataSize;
+ AudioObjectPropertyAddress property = { kAudioHardwarePropertyDevices, kAudioObjectPropertyScopeGlobal, KAUDIOOBJECTPROPERTYELEMENT };
+ OSStatus result = AudioObjectGetPropertyDataSize( kAudioObjectSystemObject, &property, 0, NULL, &dataSize );
+ if ( result != noErr ) {
+ errorText_ = "RtApiCore::probeDevices: OS-X system error getting device info!";
+ error( RTAUDIO_SYSTEM_ERROR );
+ return;
+ }
+
+ unsigned int nDevices = dataSize / sizeof( AudioDeviceID );
+ if ( nDevices == 0 ) {
+ deviceList_.clear();
+ deviceIds_.clear();
+ return;
+ }
+
+ AudioDeviceID ids[ nDevices ];
+ property.mSelector = kAudioHardwarePropertyDevices;
+ result = AudioObjectGetPropertyData( kAudioObjectSystemObject, &property, 0, NULL, &dataSize, (void *) &ids );
+ if ( result != noErr ) {
+ errorText_ = "RtApiCore::probeDevices: OS-X system error getting device IDs.";
+ error( RTAUDIO_SYSTEM_ERROR );
+ return;
+ }
+
+ // Fill or update the deviceList_ and also save a corresponding list of Ids.
+ for ( unsigned int n=0; n<nDevices; n++ ) {
+ if ( std::find( deviceIds_.begin(), deviceIds_.end(), ids[n] ) != deviceIds_.end() ) {
+ continue; // We already have this device.
+ }
+ else { // There is a new device to probe.
+ RtAudio::DeviceInfo info;
+ if ( probeDeviceInfo( ids[n], info ) == false ) continue; // ignore if probe fails
+ deviceIds_.push_back( ids[n] );
+ info.ID = currentDeviceId_++; // arbitrary internal device ID
+ deviceList_.push_back( info );
+ // We could set a property listener here for each device to know
+ // if it is removed. However, we cannot detect (AFAIK) when a new
+ // device is plugged in. If we cannot detect BOTH cases, I'm not
+ // going to bother with only the one.
+ }
+ }
+
+ // Remove any devices left in the list that are no longer available.
+ unsigned int m;
+ for ( std::vector<AudioDeviceID>::iterator it=deviceIds_.begin(); it!=deviceIds_.end(); ) {
+ for ( m=0; m<nDevices; m++ ) {
+ if ( ids[m] == *it ) {
+ ++it;
+ break;
+ }
+ }
+ if ( m == nDevices ) { // not found so remove it from our two lists
+ it = deviceIds_.erase(it);
+ deviceList_.erase( deviceList_.begin() + distance(deviceIds_.begin(), it ) );
+ }
+ }
+
+ // Get default devices and set flags in deviceList_.
+ AudioDeviceID defaultOutputId, defaultInputId;
+ dataSize = sizeof( AudioDeviceID );
+ property.mSelector = kAudioHardwarePropertyDefaultOutputDevice;
+ result = AudioObjectGetPropertyData( kAudioObjectSystemObject, &property, 0, NULL, &dataSize, &defaultOutputId );
+ if ( result != noErr ) {
+ errorText_ = "RtApiCore::probeDeviceInfo: OS-X system error getting default output device.";
+ error( RTAUDIO_WARNING );
+ defaultOutputId = 0;
+ }
+
+ property.mSelector = kAudioHardwarePropertyDefaultInputDevice;
+ result = AudioObjectGetPropertyData( kAudioObjectSystemObject, &property, 0, NULL, &dataSize, &defaultInputId );
+ if ( result != noErr ) {
+ errorText_ = "RtApiCore::probeDeviceInfo: OS-X system error getting default input device.";
+ error( RTAUDIO_WARNING );
+ defaultInputId = 0;
+ }
+
+ for ( m=0; m<deviceList_.size(); m++ ) {
+ if ( deviceIds_[m] == defaultOutputId )
+ deviceList_[m].isDefaultOutput = true;
+ else
+ deviceList_[m].isDefaultOutput = false;
+ if ( deviceIds_[m] == defaultInputId )
+ deviceList_[m].isDefaultInput = true;
+ else
+ deviceList_[m].isDefaultInput = false;
+ }
+}
+
+bool RtApiCore :: probeDeviceInfo( AudioDeviceID id, RtAudio::DeviceInfo& info )
+{
+ // Get the device name.
+ info.name.erase();
+ CFStringRef cfname;
+ UInt32 dataSize = sizeof( CFStringRef );
+ AudioObjectPropertyAddress property = { kAudioObjectPropertyManufacturer,
+ kAudioObjectPropertyScopeGlobal,
+ KAUDIOOBJECTPROPERTYELEMENT };
+ OSStatus 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( RTAUDIO_WARNING );
+ return false;
+ }
+
+ long 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( RTAUDIO_WARNING );
+ return false;
+ }
+
+ 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;
+ dataSize = 0;
+ result = AudioObjectGetPropertyDataSize( id, &property, 0, NULL, &dataSize );
+ if ( result != noErr || dataSize == 0 ) {
+ errorStream_ << "RtApiCore::probeDeviceInfo: system error (" << getErrorCode( result ) << ") getting output stream configuration info for device (" << info.name << ").";
+ errorText_ = errorStream_.str();
+ error( RTAUDIO_WARNING );
+ return false;
+ }
+
+ // Allocate the AudioBufferList.
+ bufferList = (AudioBufferList *) malloc( dataSize );
+ if ( bufferList == NULL ) {
+ errorText_ = "RtApiCore::probeDeviceInfo: memory error allocating output AudioBufferList.";
+ error( RTAUDIO_WARNING );
+ return false;
+ }
+
+ result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, bufferList );
+ if ( result != noErr || dataSize == 0 ) {
+ free( bufferList );
+ errorStream_ << "RtApiCore::probeDeviceInfo: system error (" << getErrorCode( result ) << ") getting output stream configuration for device (" << info.name << ").";
+ errorText_ = errorStream_.str();
+ error( RTAUDIO_WARNING );
+ return false;
+ }
+
+ // 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::probeDeviceInfo: system error (" << getErrorCode( result ) << ") getting input stream configuration info for device (" << info.name << ").";
+ errorText_ = errorStream_.str();
+ error( RTAUDIO_WARNING );
+ return false;
+ }
+
+ // Allocate the AudioBufferList.
+ bufferList = (AudioBufferList *) malloc( dataSize );
+ if ( bufferList == NULL ) {
+ errorText_ = "RtApiCore::probeDeviceInfo: memory error allocating input AudioBufferList.";
+ error( RTAUDIO_WARNING );
+ return false;
+ }
+
+ result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, bufferList );
+ if (result != noErr || dataSize == 0) {
+ free( bufferList );
+ errorStream_ << "RtApiCore::probeDeviceInfo: system error (" << getErrorCode( result ) << ") getting input stream configuration for device (" << info.name << ").";
+ errorText_ = errorStream_.str();
+ error( RTAUDIO_WARNING );
+ return false;
+ }
+
+ // 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::probeDeviceInfo: system error (" << getErrorCode( result ) << ") getting sample rate info.";
+ errorText_ = errorStream_.str();
+ error( RTAUDIO_WARNING );
+ return false;
+ }
+
+ UInt32 nRanges = dataSize / sizeof( AudioValueRange );
+ AudioValueRange rangeList[ nRanges ];
+ result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, &rangeList );
+ if ( result != kAudioHardwareNoError ) {
+ errorStream_ << "RtApiCore::probeDeviceInfo: system error (" << getErrorCode( result ) << ") getting sample rates.";
+ errorText_ = errorStream_.str();
+ error( RTAUDIO_WARNING );
+ return false;
+ }
+
+ // 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 (" << info.name << ").";
+ errorText_ = errorStream_.str();
+ error( RTAUDIO_WARNING );
+ return false;
+ }
+
+ // Probe the currently configured sample rate
+ Float64 nominalRate;
+ dataSize = sizeof( Float64 );
+ property.mSelector = kAudioDevicePropertyNominalSampleRate;
+ result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, &nominalRate );
+ if ( result == noErr ) info.currentSampleRate = (unsigned int) nominalRate;
+
+ // 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;
+
+ return true;
+}
+
+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;
+ if(info == NULL || info->object == NULL)
+ return kAudioHardwareUnspecifiedError;
+
+ 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;
+}
+
+bool RtApiCore :: probeDeviceOpen( unsigned int deviceId, StreamMode mode, unsigned int channels,
+ unsigned int firstChannel, unsigned int sampleRate,
+ RtAudioFormat format, unsigned int *bufferSize,
+ RtAudio::StreamOptions *options )
+{
+ AudioDeviceID id = 0;
+ for ( unsigned int m=0; m<deviceList_.size(); m++ ) {
+ if ( deviceList_[m].ID == deviceId ) {
+ id = deviceIds_[m];
+ break;
+ }
+ }
+
+ if ( id == 0 ) {
+ errorText_ = "RtApiCore::probeDeviceOpen: the device ID was not found!";
+ return FAILURE;
+ }
+
+ AudioObjectPropertyAddress property = { kAudioHardwarePropertyDevices,
+ kAudioDevicePropertyScopeOutput,
+ KAUDIOOBJECTPROPERTYELEMENT };
+
+ // Setup for stream mode.
+ if ( mode == INPUT ) {
+ property.mScope = kAudioDevicePropertyScopeInput;
+ }
+
+ // Get the stream "configuration".
+ AudioBufferList *bufferList = nil;
+ UInt32 dataSize = 0;
+ property.mSelector = kAudioDevicePropertyStreamConfiguration;
+ OSStatus 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 (" << deviceId << ").";
+ 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 (" << deviceId << ").";
+ 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 (" << deviceId << ") does not support the requested channel count.";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
+
+ // Look for a single stream meeting our needs.
+ UInt32 firstStream = 0, 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 (" << deviceId << ").";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
+
+ if ( bufferRange.mMinimum > *bufferSize ) *bufferSize = (unsigned int) bufferRange.mMinimum;
+ else if ( bufferRange.mMaximum < *bufferSize ) *bufferSize = (unsigned int) bufferRange.mMaximum;
+ if ( options && options->flags & RTAUDIO_MINIMIZE_LATENCY ) *bufferSize = (unsigned int) 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 (" << deviceId << ").";
+ 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 (" << deviceId << ").";
+ 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 try to change the sample rate if off by more than 1 Hz.
+ if ( fabs( nominalRate - (double)sampleRate ) > 1.0 ) {
+
+ nominalRate = (Float64) sampleRate;
+ result = AudioObjectSetPropertyData( id, &property, 0, NULL, dataSize, &nominalRate );
+ if ( result != noErr ) {
+ errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") setting sample rate for device (" << deviceId << ").";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
+
+ // Now wait until the reported nominal rate is what we just set.
+ UInt32 microCounter = 0;
+ Float64 reportedRate = 0.0;
+ while ( reportedRate != nominalRate ) {
+ microCounter += 5000;
+ if ( microCounter > 2000000 ) break;
+ usleep( 5000 );
+ result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, &reportedRate );
+ }
+
+ if ( microCounter > 2000000 ) {
+ errorStream_ << "RtApiCore::probeDeviceOpen: timeout waiting for sample rate update for device (" << deviceId << ").";
+ 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 (" << deviceId << ").";
+ 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 (" << deviceId << ").";
+ 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 (" << deviceId << ").";
+ 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 (" << deviceId << ").";
+ 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 (" << deviceId << ").";
+ errorText_ = errorStream_.str();
+ error( RTAUDIO_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 );
+ 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_.deviceId[mode] = deviceId;
+ 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_.deviceId[0] == deviceId )
+ // Only one callback procedure and property listener 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 (" << deviceId << ").";
+ 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 );
+ if ( result != noErr ) {
+ errorStream_ << "RtApiCore::probeDeviceOpen: system error setting xrun listener for device (" << deviceId << ").";
+ errorText_ = errorStream_.str();
+ goto error;
+ }
+ handle->xrunListenerAdded[mode] = true;
+
+ // Setup a listener to detect a possible device disconnect.
+ property.mSelector = kAudioDevicePropertyDeviceIsAlive;
+ result = AudioObjectAddPropertyListener( id , &property, streamDisconnectListener, (void *) &stream_.callbackInfo );
+ if ( result != noErr ) {
+ AudioObjectRemovePropertyListener( id, &property, xrunListener, (void *) handle );
+ errorStream_ << "RtApiCore::probeDeviceOpen: system error setting disconnect listener for device (" << deviceId << ").";
+ errorText_ = errorStream_.str();
+ goto error;
+ }
+ handle->disconnectListenerAdded[mode] = true;
+ }
+
+ return SUCCESS;
+
+ error:
+ closeStream(); // this should safely clear out procedures, listeners and memory, even for duplex stream
+ return FAILURE;
+}
+
+void RtApiCore :: closeStream( void )
+{
+ if ( stream_.state == STREAM_CLOSED ) {
+ errorText_ = "RtApiCore::closeStream(): no open stream to close!";
+ error( RTAUDIO_WARNING );
+ return;
+ }
+
+ CoreHandle *handle = (CoreHandle *) stream_.apiHandle;
+ if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
+ if ( handle ) {
+ AudioObjectPropertyAddress property = { kAudioHardwarePropertyDevices,
+ kAudioObjectPropertyScopeGlobal,
+ KAUDIOOBJECTPROPERTYELEMENT };
+ if ( handle->xrunListenerAdded[0] ) {
+ property.mSelector = kAudioDeviceProcessorOverload;
+ if (AudioObjectRemovePropertyListener( handle->id[0], &property, xrunListener, (void *) handle ) != noErr) {
+ errorText_ = "RtApiCore::closeStream(): error removing xrun property listener!";
+ error( RTAUDIO_WARNING );
+ }
+ }
+ if ( handle->disconnectListenerAdded[0] ) {
+ property.mSelector = kAudioDevicePropertyDeviceIsAlive;
+ if (AudioObjectRemovePropertyListener( handle->id[0], &property, streamDisconnectListener, (void *) &stream_.callbackInfo ) != noErr) {
+ errorText_ = "RtApiCore::closeStream(): error removing disconnect property listener!";
+ error( RTAUDIO_WARNING );
+ }
+ }
+
+#if defined( MAC_OS_X_VERSION_10_5 ) && ( MAC_OS_X_VERSION_MIN_REQUIRED >= MAC_OS_X_VERSION_10_5 )
+ if ( handle->procId[0] ) {
+ if ( stream_.state == STREAM_RUNNING )
+ AudioDeviceStop( handle->id[0], handle->procId[0] );
+ AudioDeviceDestroyIOProcID( handle->id[0], handle->procId[0] );
+ }
+#else // deprecated behaviour
+ if ( stream_.state == STREAM_RUNNING )
+ AudioDeviceStop( handle->id[0], callbackHandler );
+ AudioDeviceRemoveIOProc( handle->id[0], callbackHandler );
+#endif
+ }
+ }
+
+ if ( stream_.mode == INPUT || ( stream_.mode == DUPLEX && stream_.deviceId[0] != stream_.deviceId[1] ) ) {
+ if ( handle ) {
+ AudioObjectPropertyAddress property = { kAudioHardwarePropertyDevices,
+ kAudioObjectPropertyScopeGlobal,
+ KAUDIOOBJECTPROPERTYELEMENT };
+
+ if ( handle->xrunListenerAdded[1] ) {
+ property.mSelector = kAudioDeviceProcessorOverload;
+ if (AudioObjectRemovePropertyListener( handle->id[1], &property, xrunListener, (void *) handle ) != noErr) {
+ errorText_ = "RtApiCore::closeStream(): error removing xrun property listener!";
+ error( RTAUDIO_WARNING );
+ }
+ }
+
+ if ( handle->disconnectListenerAdded[0] ) {
+ property.mSelector = kAudioDevicePropertyDeviceIsAlive;
+ if (AudioObjectRemovePropertyListener( handle->id[1], &property, streamDisconnectListener, (void *) &stream_.callbackInfo ) != noErr) {
+ errorText_ = "RtApiCore::closeStream(): error removing disconnect property listener!";
+ error( RTAUDIO_WARNING );
+ }
+ }
+
+#if defined( MAC_OS_X_VERSION_10_5 ) && ( MAC_OS_X_VERSION_MIN_REQUIRED >= MAC_OS_X_VERSION_10_5 )
+ if ( handle->procId[1] ) {
+ if ( stream_.state == STREAM_RUNNING )
+ AudioDeviceStop( handle->id[1], handle->procId[1] );
+ AudioDeviceDestroyIOProcID( handle->id[1], handle->procId[1] );
+ }
+#else // deprecated behaviour
+ if ( stream_.state == STREAM_RUNNING )
+ AudioDeviceStop( handle->id[1], callbackHandler );
+ 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_signal( &handle->condition ); // signal condition variable in case stopStream is blocked
+ pthread_cond_destroy( &handle->condition );
+ delete handle;
+ stream_.apiHandle = 0;
+
+ CallbackInfo *info = (CallbackInfo *) &stream_.callbackInfo;
+ if ( info->deviceDisconnected ) {
+ errorText_ = "RtApiCore: the stream device was disconnected (and closed)!";
+ error( RTAUDIO_DEVICE_DISCONNECT );
+ }
+
+ clearStreamInfo();
+}
+
+RtAudioErrorType RtApiCore :: startStream( void )
+{
+ if ( stream_.state != STREAM_STOPPED ) {
+ if ( stream_.state == STREAM_RUNNING )
+ errorText_ = "RtApiCore::startStream(): the stream is already running!";
+ else if ( stream_.state == STREAM_STOPPING || stream_.state == STREAM_CLOSED )
+ errorText_ = "RtApiCore::startStream(): the stream is stopping or closed!";
+ return error( RTAUDIO_WARNING );
+ }
+
+ /*
+ #if defined( HAVE_GETTIMEOFDAY )
+ gettimeofday( &stream_.lastTickTimestamp, NULL );
+ #endif
+ */
+
+ OSStatus result = noErr;
+ CoreHandle *handle = (CoreHandle *) stream_.apiHandle;
+ if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
+
+#if defined( MAC_OS_X_VERSION_10_5 ) && ( MAC_OS_X_VERSION_MIN_REQUIRED >= MAC_OS_X_VERSION_10_5 )
+ result = AudioDeviceStart( handle->id[0], handle->procId[0] );
+#else // deprecated behaviour
+ result = AudioDeviceStart( handle->id[0], callbackHandler );
+#endif
+ if ( result != noErr ) {
+ errorStream_ << "RtApiCore::startStream: system error (" << getErrorCode( result ) << ") starting callback procedure on device (" << stream_.deviceId[0] << ").";
+ errorText_ = errorStream_.str();
+ goto unlock;
+ }
+ }
+
+ if ( stream_.mode == INPUT ||
+ ( stream_.mode == DUPLEX && stream_.deviceId[0] != stream_.deviceId[1] ) ) {
+
+ // Clear user input buffer
+ unsigned long bufferBytes;
+ bufferBytes = stream_.nUserChannels[1] * stream_.bufferSize * formatBytes( stream_.userFormat );
+ memset( stream_.userBuffer[1], 0, bufferBytes * sizeof(char) );
+
+#if defined( MAC_OS_X_VERSION_10_5 ) && ( MAC_OS_X_VERSION_MIN_REQUIRED >= MAC_OS_X_VERSION_10_5 )
+ result = AudioDeviceStart( handle->id[1], handle->procId[1] );
+#else // deprecated behaviour
+ result = AudioDeviceStart( handle->id[1], callbackHandler );
+#endif
+ if ( result != noErr ) {
+ errorStream_ << "RtApiCore::startStream: system error starting input callback procedure on device (" << stream_.deviceId[1] << ").";
+ errorText_ = errorStream_.str();
+ goto unlock;
+ }
+ }
+
+ handle->drainCounter = 0;
+ handle->internalDrain = false;
+ stream_.state = STREAM_RUNNING;
+
+ unlock:
+ if ( result == noErr ) return RTAUDIO_NO_ERROR;
+ return error( RTAUDIO_SYSTEM_ERROR );
+}
+
+RtAudioErrorType RtApiCore :: stopStream( void )
+{
+ if ( stream_.state != STREAM_RUNNING && stream_.state != STREAM_STOPPING ) {
+ if ( stream_.state == STREAM_STOPPED )
+ errorText_ = "RtApiCore::stopStream(): the stream is already stopped!";
+ else if ( stream_.state == STREAM_CLOSED )
+ errorText_ = "RtApiCore::stopStream(): the stream is closed!";
+ return error( RTAUDIO_WARNING );
+ }
+
+ 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
+ }
+
+#if defined( MAC_OS_X_VERSION_10_5 ) && ( MAC_OS_X_VERSION_MIN_REQUIRED >= MAC_OS_X_VERSION_10_5 )
+ result = AudioDeviceStop( handle->id[0], handle->procId[0] );
+#else // deprecated behaviour
+ result = AudioDeviceStop( handle->id[0], callbackHandler );
+#endif
+ if ( result != noErr ) {
+ errorStream_ << "RtApiCore::stopStream: system error (" << getErrorCode( result ) << ") stopping callback procedure on device (" << stream_.deviceId[0] << ").";
+ errorText_ = errorStream_.str();
+ goto unlock;
+ }
+ }
+
+ if ( stream_.mode == INPUT || ( stream_.mode == DUPLEX && stream_.deviceId[0] != stream_.deviceId[1] ) ) {
+#if defined( MAC_OS_X_VERSION_10_5 ) && ( MAC_OS_X_VERSION_MIN_REQUIRED >= MAC_OS_X_VERSION_10_5 )
+ result = AudioDeviceStop( handle->id[1], handle->procId[1] );
+#else // deprecated behaviour
+ result = AudioDeviceStop( handle->id[1], callbackHandler );
+#endif
+ if ( result != noErr ) {
+ errorStream_ << "RtApiCore::stopStream: system error (" << getErrorCode( result ) << ") stopping input callback procedure on device (" << stream_.deviceId[1] << ").";
+ errorText_ = errorStream_.str();
+ goto unlock;
+ }
+ }
+
+ stream_.state = STREAM_STOPPED;
+
+ unlock:
+ if ( result == noErr ) return RTAUDIO_NO_ERROR;
+ return error( RTAUDIO_SYSTEM_ERROR );
+}
+
+RtAudioErrorType RtApiCore :: abortStream( void )
+{
+ if ( stream_.state != STREAM_RUNNING ) {
+ if ( stream_.state == STREAM_STOPPED )
+ errorText_ = "RtApiCore::abortStream(): the stream is already stopped!";
+ else if ( stream_.state == STREAM_STOPPING || stream_.state == STREAM_CLOSED )
+ errorText_ = "RtApiCore::abortStream(): the stream is stopping or closed!";
+ return error( RTAUDIO_WARNING );
+ }
+
+ CoreHandle *handle = (CoreHandle *) stream_.apiHandle;
+ handle->drainCounter = 2;
+
+ stream_.state = STREAM_STOPPING;
+ return 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( RTAUDIO_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 ) {
+ 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:
+
+ // Make sure to only tick duplex stream time once if using two devices
+ if ( stream_.mode == DUPLEX ) {
+ if ( handle->id[0] == handle->id[1] ) // same device, only one callback
+ RtApi::tickStreamTime();
+ else if ( deviceId == handle->id[0] )
+ RtApi::tickStreamTime(); // two devices, only tick on the output callback
+ } else
+ RtApi::tickStreamTime(); // input or output stream only
+
+ 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 and
+// Windows. 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, while ports correspond to device
+// channels. 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 <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; }
+};
+
+std::string escapeJackPortRegex(std::string &str)
+{
+ const std::string need_escaping = "()[]{}*+?$^.|\\";
+ std::string escaped_string;
+ for (auto c : str)
+ {
+ if (need_escaping.find(c) != std::string::npos)
+ escaped_string.push_back('\\');
+
+ escaped_string.push_back(c);
+ }
+ return escaped_string;
+}
+
+#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();
+}
+
+void RtApiJack :: probeDevices( void )
+{
+ // See list of required functionality in RtApi::probeDevices().
+
+ // 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( "RtApiJackProbe", options, status );
+ if ( client == 0 ) {
+ deviceList_.clear(); // in case the server is shutdown after a previous successful probe
+ errorText_ = "RtApiJack::probeDevices: Jack server not found or connection error!";
+ //error( RTAUDIO_SYSTEM_ERROR );
+ error( RTAUDIO_WARNING );
+ return;
+ }
+
+ const char **ports;
+ std::string port, previousPort;
+ unsigned int nChannels = 0, nDevices = 0;
+ std::vector<std::string> portNames;
+ ports = jack_get_ports( client, NULL, JACK_DEFAULT_AUDIO_TYPE, 0 );
+ if ( ports ) {
+ // Parse the port names up to the first colon (:).
+ size_t iColon = 0;
+ do {
+ port = (char *) ports[ nChannels ];
+ iColon = port.find(":");
+ if ( iColon != std::string::npos ) {
+ port = port.substr( 0, iColon );
+ if ( port != previousPort ) {
+ portNames.push_back( port );
+ nDevices++;
+ previousPort = port;
+ }
+ }
+ } while ( ports[++nChannels] );
+ free( ports );
+ }
+
+ // Fill or update the deviceList_.
+ unsigned int m, n;
+ for ( n=0; n<nDevices; n++ ) {
+ for ( m=0; m<deviceList_.size(); m++ ) {
+ if ( deviceList_[m].name == portNames[n] )
+ break; // We already have this device.
+ }
+ if ( m == deviceList_.size() ) { // new device
+ RtAudio::DeviceInfo info;
+ info.name = portNames[n];
+ if ( probeDeviceInfo( info, client ) == false ) continue; // ignore if probe fails
+ info.ID = currentDeviceId_++; // arbitrary internal device ID
+ deviceList_.push_back( info );
+ // A callback can be registered in Jack to be notified about client
+ // (dis)connections. However, this can only be done with an open client,
+ // so unless we want to keep a special client open all the time, this
+ // would only report (dis)connections when a stream is open. I'm not
+ // going to bother for the moment.
+ }
+ }
+
+ // Remove any devices left in the list that are no longer available.
+ for ( std::vector<RtAudio::DeviceInfo>::iterator it=deviceList_.begin(); it!=deviceList_.end(); ) {
+ for ( m=0; m<portNames.size(); m++ ) {
+ if ( (*it).name == portNames[m] ) {
+ ++it;
+ break;
+ }
+ }
+ if ( m == portNames.size() ) // not found so remove it from our list
+ it = deviceList_.erase( it );
+ }
+
+ jack_client_close( client );
+
+ if ( nDevices == 0 ) {
+ deviceList_.clear();
+ return;
+ }
+
+ // Jack doesn't provide default devices so call the getDefault
+ // functions, which will set the first available input and output
+ // devices as the defaults.
+ getDefaultInputDevice();
+ getDefaultOutputDevice();
+}
+
+bool RtApiJack :: probeDeviceInfo( RtAudio::DeviceInfo& info, jack_client_t *client )
+{
+ // 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;
+ const char **ports = jack_get_ports( client, escapeJackPortRegex(info.name).c_str(), JACK_DEFAULT_AUDIO_TYPE, JackPortIsInput );
+ if ( ports ) {
+ while ( ports[ nChannels ] ) nChannels++;
+ free( ports );
+ info.outputChannels = nChannels;
+ }
+
+ // Jack "output ports" equal RtAudio input channels.
+ nChannels = 0;
+ ports = jack_get_ports( client, escapeJackPortRegex(info.name).c_str(), JACK_DEFAULT_AUDIO_TYPE, JackPortIsOutput );
+ if ( ports ) {
+ while ( ports[ nChannels ] ) nChannels++;
+ free( ports );
+ info.inputChannels = nChannels;
+ }
+
+ if ( info.outputChannels == 0 && info.inputChannels == 0 ) {
+ jack_client_close(client);
+ errorText_ = "RtApiJack::getDeviceInfo: error determining Jack input/output channels!";
+ error( RTAUDIO_WARNING );
+ return false;
+ }
+
+ // 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;
+
+ return true;
+}
+
+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;
+
+ info->deviceDisconnected = true;
+ object->closeStream();
+ pthread_exit( NULL );
+}
+
+/*
+// Could be used to catch client connections but requires open client.
+static void jackClientChange( const char *name, int registered, void *infoPointer )
+{
+ std::cout << "in jackClientChange, name = " << name << ", registered = " << registered << std::endl;
+}
+*/
+
+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 );
+}
+
+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 deviceId, 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( RTAUDIO_WARNING );
+ return FAILURE;
+ }
+ }
+ else {
+ // The handle must have been created on an earlier pass.
+ client = handle->client;
+ }
+
+ std::string deviceName;
+ for ( unsigned int m=0; m<deviceList_.size(); m++ ) {
+ if ( deviceList_[m].ID == deviceId ) {
+ deviceName = deviceList_[m].name;
+ break;
+ }
+ }
+
+ if ( deviceName.empty() ) {
+ errorText_ = "RtApiJack::probeDeviceOpen: device ID is invalid!";
+ return FAILURE;
+ }
+
+ unsigned long flag = JackPortIsInput;
+ if ( mode == INPUT ) flag = JackPortIsOutput;
+
+ const char **ports;
+ if ( ! (options && (options->flags & RTAUDIO_JACK_DONT_CONNECT)) ) {
+ // Count the available ports containing the client name as device
+ // channels. Jack "input ports" equal RtAudio output channels.
+ unsigned int nChannels = 0;
+ ports = jack_get_ports( client, escapeJackPortRegex(deviceName).c_str(), JACK_DEFAULT_AUDIO_TYPE, flag );
+ if ( ports ) {
+ while ( ports[ nChannels ] ) nChannels++;
+ free( ports );
+ }
+ // Compare the jack ports for specified client to the requested number of channels.
+ if ( nChannels < (channels + firstChannel) ) {
+ errorStream_ << "RtApiJack::probeDeviceOpen: requested number of channels (" << channels << ") + offset (" << firstChannel << ") not found for specified device (" << 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, escapeJackPortRegex(deviceName).c_str(), JACK_DEFAULT_AUDIO_TYPE, flag );
+ if ( ports[ firstChannel ] ) {
+ // Added by Ge Wang
+ jack_latency_callback_mode_t cbmode = (mode == INPUT ? JackCaptureLatency : JackPlaybackLatency);
+ // the range (usually the min and max are equal)
+ jack_latency_range_t latrange; latrange.min = latrange.max = 0;
+ // get the latency range
+ jack_port_get_latency_range( jack_port_by_name( client, ports[firstChannel] ), cbmode, &latrange );
+ // be optimistic, use the min!
+ stream_.latency[mode] = latrange.min;
+ //stream_.latency[mode] = jack_port_get_latency( jack_port_by_name( client, ports[ firstChannel ] ) );
+ }
+ free( ports );
+
+ // The jack server always uses 32-bit floating-point data.
+ stream_.deviceFormat[mode] = RTAUDIO_FLOAT32;
+ stream_.userFormat = format;
+
+ if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) stream_.userInterleaved = false;
+ else stream_.userInterleaved = true;
+
+ // Jack always uses non-interleaved buffers.
+ stream_.deviceInterleaved[mode] = false;
+
+ // Jack always provides host byte-ordered data.
+ stream_.doByteSwap[mode] = false;
+
+ // Get the buffer size. The buffer size and number of buffers
+ // (periods) is set when the jack server is started.
+ stream_.bufferSize = (int) jack_get_buffer_size( client );
+ *bufferSize = stream_.bufferSize;
+
+ stream_.nDeviceChannels[mode] = channels;
+ stream_.nUserChannels[mode] = channels;
+
+ // Set flags for buffer conversion.
+ stream_.doConvertBuffer[mode] = false;
+ if ( stream_.userFormat != stream_.deviceFormat[mode] )
+ stream_.doConvertBuffer[mode] = true;
+ if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] &&
+ stream_.nUserChannels[mode] > 1 )
+ stream_.doConvertBuffer[mode] = true;
+
+ // Allocate our JackHandle structure for the stream.
+ if ( handle == 0 ) {
+ try {
+ handle = new JackHandle;
+ }
+ catch ( std::bad_alloc& ) {
+ errorText_ = "RtApiJack::probeDeviceOpen: error allocating JackHandle memory.";
+ goto error;
+ }
+
+ if ( pthread_cond_init(&handle->condition, NULL) ) {
+ errorText_ = "RtApiJack::probeDeviceOpen: error initializing pthread condition variable.";
+ goto error;
+ }
+ stream_.apiHandle = (void *) handle;
+ handle->client = client;
+ }
+ handle->deviceName[mode] = deviceName;
+
+ // Allocate necessary internal buffers.
+ unsigned long bufferBytes;
+ bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat );
+ stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 );
+ if ( stream_.userBuffer[mode] == NULL ) {
+ errorText_ = "RtApiJack::probeDeviceOpen: error allocating user buffer memory.";
+ goto error;
+ }
+
+ if ( stream_.doConvertBuffer[mode] ) {
+
+ bool makeBuffer = true;
+ if ( mode == OUTPUT )
+ bufferBytes = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] );
+ else { // mode == INPUT
+ bufferBytes = stream_.nDeviceChannels[1] * formatBytes( stream_.deviceFormat[1] );
+ if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) {
+ unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes(stream_.deviceFormat[0]);
+ if ( bufferBytes < bytesOut ) makeBuffer = false;
+ }
+ }
+
+ if ( makeBuffer ) {
+ bufferBytes *= *bufferSize;
+ if ( stream_.deviceBuffer ) free( stream_.deviceBuffer );
+ stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 );
+ if ( stream_.deviceBuffer == NULL ) {
+ errorText_ = "RtApiJack::probeDeviceOpen: error allocating device buffer memory.";
+ goto error;
+ }
+ }
+ }
+
+ // Allocate memory for the Jack ports (channels) identifiers.
+ handle->ports[mode] = (jack_port_t **) malloc ( sizeof (jack_port_t *) * channels );
+ if ( handle->ports[mode] == NULL ) {
+ errorText_ = "RtApiJack::probeDeviceOpen: error allocating port memory.";
+ goto error;
+ }
+
+ stream_.channelOffset[mode] = firstChannel;
+ stream_.state = STREAM_STOPPED;
+ stream_.callbackInfo.object = (void *) this;
+
+ if ( stream_.mode == OUTPUT && mode == INPUT )
+ // We had already set up the stream for output.
+ stream_.mode = DUPLEX;
+ else {
+ stream_.mode = mode;
+ jack_set_process_callback( handle->client, jackCallbackHandler, (void *) &stream_.callbackInfo );
+ jack_set_xrun_callback( handle->client, jackXrun, (void *) &stream_.apiHandle );
+ jack_on_shutdown( handle->client, jackShutdown, (void *) &stream_.callbackInfo );
+ //jack_set_client_registration_callback( handle->client, jackClientChange, (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( RTAUDIO_WARNING );
+ return;
+ }
+
+ JackHandle *handle = (JackHandle *) stream_.apiHandle;
+ if ( handle ) {
+ if ( stream_.state == STREAM_RUNNING )
+ jack_deactivate( handle->client );
+
+ if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
+ for ( unsigned int i=0; i<stream_.nUserChannels[0]; i++ )
+ jack_port_unregister( handle->client, handle->ports[0][i] );
+ }
+ if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {
+ for ( unsigned int i=0; i<stream_.nUserChannels[1]; i++ )
+ jack_port_unregister( handle->client, handle->ports[1][i] );
+ }
+ jack_client_close( handle->client );
+
+ 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;
+ }
+
+ CallbackInfo *info = (CallbackInfo *) &stream_.callbackInfo;
+ if ( info->deviceDisconnected ) {
+ errorText_ = "RtApiJack: the Jack server is shutting down this client ... stream stopped and closed!";
+ error( RTAUDIO_DEVICE_DISCONNECT );
+ }
+
+ 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;
+ }
+
+ clearStreamInfo();
+}
+
+RtAudioErrorType RtApiJack :: startStream( void )
+{
+ if ( stream_.state != STREAM_STOPPED ) {
+ if ( stream_.state == STREAM_RUNNING )
+ errorText_ = "RtApiJack::startStream(): the stream is already running!";
+ else if ( stream_.state == STREAM_STOPPING || stream_.state == STREAM_CLOSED )
+ errorText_ = "RtApiJack::startStream(): the stream is stopping or closed!";
+ return error( RTAUDIO_WARNING );
+ }
+
+ /*
+ #if defined( HAVE_GETTIMEOFDAY )
+ gettimeofday( &stream_.lastTickTimestamp, NULL );
+ #endif
+ */
+
+ 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) ) {
+ ports = jack_get_ports( handle->client, escapeJackPortRegex(handle->deviceName[0]).c_str(), JACK_DEFAULT_AUDIO_TYPE, JackPortIsInput);
+ if ( ports == NULL) {
+ errorText_ = "RtApiJack::startStream(): error determining available JACK input ports!";
+ goto unlock;
+ }
+
+ // Now make the port connections. Since RtAudio wasn't designed to
+ // allow the user to select particular channels of a device, we'll
+ // just open the first "nChannels" ports with offset.
+ for ( unsigned int i=0; i<stream_.nUserChannels[0]; i++ ) {
+ result = 1;
+ if ( ports[ stream_.channelOffset[0] + i ] )
+ result = jack_connect( handle->client, jack_port_name( handle->ports[0][i] ), ports[ stream_.channelOffset[0] + i ] );
+ if ( result ) {
+ free( ports );
+ errorText_ = "RtApiJack::startStream(): error connecting output ports!";
+ goto unlock;
+ }
+ }
+ free(ports);
+ }
+
+ if ( shouldAutoconnect_ && (stream_.mode == INPUT || stream_.mode == DUPLEX) ) {
+ ports = jack_get_ports( handle->client, escapeJackPortRegex(handle->deviceName[1]).c_str(), JACK_DEFAULT_AUDIO_TYPE, JackPortIsOutput );
+ if ( ports == NULL) {
+ errorText_ = "RtApiJack::startStream(): error determining available JACK output ports!";
+ goto unlock;
+ }
+
+ // Now make the port connections. See note above.
+ for ( unsigned int i=0; i<stream_.nUserChannels[1]; i++ ) {
+ result = 1;
+ if ( ports[ stream_.channelOffset[1] + i ] )
+ result = jack_connect( handle->client, ports[ stream_.channelOffset[1] + i ], jack_port_name( handle->ports[1][i] ) );
+ if ( result ) {
+ free( ports );
+ errorText_ = "RtApiJack::startStream(): error connecting input ports!";
+ goto unlock;
+ }
+ }
+ free(ports);
+ }
+
+ handle->drainCounter = 0;
+ handle->internalDrain = false;
+ stream_.state = STREAM_RUNNING;
+
+ unlock:
+ if ( result == 0 ) return RTAUDIO_NO_ERROR;
+ return error( RTAUDIO_SYSTEM_ERROR );
+}
+
+RtAudioErrorType RtApiJack :: stopStream( void )
+{
+ if ( stream_.state != STREAM_RUNNING && stream_.state != STREAM_STOPPING ) {
+ if ( stream_.state == STREAM_STOPPED )
+ errorText_ = "RtApiJack::stopStream(): the stream is already stopped!";
+ else if ( stream_.state == STREAM_CLOSED )
+ errorText_ = "RtApiJack::stopStream(): the stream is closed!";
+ return error( RTAUDIO_WARNING );
+ }
+
+ 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;
+ return RTAUDIO_NO_ERROR;
+}
+
+RtAudioErrorType RtApiJack :: abortStream( void )
+{
+ if ( stream_.state != STREAM_RUNNING ) {
+ if ( stream_.state == STREAM_STOPPED )
+ errorText_ = "RtApiJack::abortStream(): the stream is already stopped!";
+ else if ( stream_.state == STREAM_STOPPING || stream_.state == STREAM_CLOSED )
+ errorText_ = "RtApiJack::abortStream(): the stream is stopping or closed!";
+ return error( RTAUDIO_WARNING );
+ }
+
+ JackHandle *handle = (JackHandle *) stream_.apiHandle;
+ handle->drainCounter = 2;
+
+ return 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_ = "RtApiJack::callbackEvent(): the stream is closed ... this shouldn't happen!";
+ error( RTAUDIO_WARNING );
+ return FAILURE;
+ }
+ if ( stream_.bufferSize != nframes ) {
+ errorText_ = "RtApiJack::callbackEvent(): the JACK buffer size has changed ... cannot process!";
+ error( RTAUDIO_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 // external call to stopStream()
+ 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 unix
+// 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;
+static bool streamOpen = false; // Tracks whether any instance of RtAudio has a stream open
+
+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 apartment. You can call
+ // CoInitialize beforehand, but it must be for apartment 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 apartment. Call CoInitializeEx(0,COINIT_APARTMENTTHREADED)";
+ error( RTAUDIO_WARNING );
+ }
+ coInitialized_ = true;
+
+ // Check whether another RtAudio instance has an ASIO stream open.
+ if ( streamOpen ) {
+ errorText_ = "RtApiAsio(): Another RtAudio ASIO stream is open, functionality may be limited.";
+ error( RTAUDIO_WARNING );
+ }
+ else
+ 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();
+}
+
+void RtApiAsio :: probeDevices( void )
+{
+ // See list of required functionality in RtApi::probeDevices().
+
+ if ( streamOpen ) {
+ errorText_ = "RtApiAsio::probeDevices: Another RtAudio ASIO stream is open, cannot probe devices.";
+ error( RTAUDIO_WARNING );
+ return;
+ }
+
+ unsigned int nDevices = drivers.asioGetNumDev();
+ if ( nDevices == 0 ) {
+ deviceList_.clear();
+ return;
+ }
+
+ char tmp[32];
+ std::vector< std::string > driverNames;
+ unsigned int n, m;
+ for ( n=0; n<nDevices; n++ ) {
+ ASIOError result = drivers.asioGetDriverName( (int) n, tmp, 32 );
+ if ( result != ASE_OK ) {
+ errorStream_ << "RtApiAsio::probeDevices: unable to get driver name (" << getAsioErrorString( result ) << ").";
+ errorText_ = errorStream_.str();
+ error( RTAUDIO_WARNING );
+ continue;
+ }
+ driverNames.push_back( tmp );
+ for ( m=0; m<deviceList_.size(); m++ ) {
+ if ( deviceList_[m].name == driverNames.back() )
+ break; // We already have this device.
+ }
+ if ( m == deviceList_.size() ) { // new device
+ RtAudio::DeviceInfo info;
+ info.name = driverNames.back();
+ if ( probeDeviceInfo( info ) == false ) continue; // ignore if probe fails
+ info.ID = currentDeviceId_++; // arbitrary internal device ID
+ deviceList_.push_back( info );
+ }
+ }
+
+ // Remove any devices left in the list that are no longer available.
+ for ( std::vector<RtAudio::DeviceInfo>::iterator it=deviceList_.begin(); it!=deviceList_.end(); ) {
+ for ( m=0; m<driverNames.size(); m++ ) {
+ if ( (*it).name == driverNames[m] ) {
+ ++it;
+ break;
+ }
+ }
+ if ( m == driverNames.size() ) // not found so remove it from our list
+ it = deviceList_.erase( it );
+ }
+
+ // Asio doesn't provide default devices so call the getDefault
+ // functions, which will set the first available input and output
+ // devices as the defaults. Don't call getDefaultXXXDevice if
+ // deviceList is empty.
+ if(deviceList_.size() > 0)
+ {
+ getDefaultInputDevice();
+ getDefaultOutputDevice();
+ }
+}
+
+bool RtApiAsio :: probeDeviceInfo( RtAudio::DeviceInfo &info )
+{
+ if ( !drivers.loadDriver( const_cast<char *>(info.name.c_str()) ) ) {
+ errorStream_ << "RtApiAsio::probeDeviceInfo: unable to load driver (" << info.name << ").";
+ errorText_ = errorStream_.str();
+ error( RTAUDIO_WARNING );
+ return false;
+ }
+
+ ASIOError result = ASIOInit( &driverInfo );
+ if ( result != ASE_OK ) {
+ drivers.removeCurrentDriver();
+ errorStream_ << "RtApiAsio::probeDeviceInfo: error (" << getAsioErrorString( result ) << ") initializing driver (" << info.name << ").";
+ errorText_ = errorStream_.str();
+ error( RTAUDIO_WARNING );
+ return false;
+ }
+
+ // Determine the device channel information.
+ long inputChannels, outputChannels;
+ result = ASIOGetChannels( &inputChannels, &outputChannels );
+ if ( result != ASE_OK ) {
+ ASIOExit();
+ drivers.removeCurrentDriver();
+ errorStream_ << "RtApiAsio::probeDeviceInfo: error (" << getAsioErrorString( result ) << ") getting channel count (" << info.name << ").";
+ errorText_ = errorStream_.str();
+ error( RTAUDIO_WARNING );
+ return false;
+ }
+
+ 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 ) {
+ ASIOExit();
+ drivers.removeCurrentDriver();
+ errorStream_ << "RtApiAsio::probeDeviceInfo: error (" << getAsioErrorString( result ) << ") getting driver channel info (" << info.name << ").";
+ errorText_ = errorStream_.str();
+ error( RTAUDIO_WARNING );
+ return false;
+ }
+
+ 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;
+
+ ASIOExit();
+ drivers.removeCurrentDriver();
+ return true;
+}
+
+static void bufferSwitch( long index, ASIOBool /*processNow*/ )
+{
+ RtApiAsio *object = (RtApiAsio *) asioCallbackInfo->object;
+ object->callbackEvent( index );
+}
+
+bool RtApiAsio :: probeDeviceOpen( unsigned int deviceId, 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_.deviceId[0] != deviceId ) {
+ errorText_ = "RtApiAsio::probeDeviceOpen: an ASIO duplex stream must use the same device for input and output!";
+ return FAILURE;
+ }
+
+ std::string driverName;
+ for ( unsigned int m=0; m<deviceList_.size(); m++ ) {
+ if ( deviceList_[m].ID == deviceId ) {
+ driverName = deviceList_[m].name;
+ break;
+ }
+ }
+
+ if ( driverName.empty() ) {
+ errorText_ = "RtApiAsio::probeDeviceOpen: device ID is invalid!";
+ return FAILURE;
+ }
+
+ // Only load the driver once for duplex stream.
+ ASIOError result;
+ if ( !isDuplexInput ) {
+ if ( streamOpen ) {
+ errorText_ = "RtApiAsio::probeDeviceOpen: Another RtAudio ASIO stream is open, cannot open more than one at a time.";
+ return FAILURE;
+ }
+ if ( !drivers.loadDriver( const_cast<char *>(driverName.c_str()) ) ) {
+ errorStream_ << "RtApiAsio::probeDeviceOpen: unable to load driver (" << driverName << ").";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
+
+ result = ASIOInit( &driverInfo );
+ if ( result != ASE_OK ) {
+ drivers.removeCurrentDriver();
+ errorStream_ << "RtApiAsio::probeDeviceOpen: error (" << getAsioErrorString( result ) << ") initializing driver (" << driverName << ").";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
+ }
+
+ 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( &currentRate );
+ 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_.deviceId[mode] = deviceId;
+ stream_.mode = isDuplexInput ? DUPLEX : mode;
+
+ // store this class instance before registering callbacks, that are going to use it
+ asioCallbackInfo = &stream_.callbackInfo;
+ stream_.callbackInfo.object = (void *) this;
+
+ // Set up the ASIO callback structure and create the ASIO data buffers.
+ asioCallbacks.bufferSwitch = &bufferSwitch;
+ asioCallbacks.sampleRateDidChange = &sampleRateChanged;
+ asioCallbacks.asioMessage = &asioMessages;
+ asioCallbacks.bufferSwitchTimeInfo = NULL;
+ result = ASIOCreateBuffers( handle->bufferInfos, nChannels, stream_.bufferSize, &asioCallbacks );
+ if ( result != ASE_OK ) {
+ // Standard method failed. This can happen with strict/misbehaving drivers that return valid buffer size ranges
+ // but only accept the preferred buffer size as parameter for ASIOCreateBuffers (e.g. Creative's ASIO driver).
+ // In that case, let's be naïve and try that instead.
+ *bufferSize = preferSize;
+ stream_.bufferSize = *bufferSize;
+ result = ASIOCreateBuffers( handle->bufferInfos, nChannels, stream_.bufferSize, &asioCallbacks );
+ }
+
+ if ( result != ASE_OK ) {
+ errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error (" << getAsioErrorString( result ) << ") creating buffers.";
+ errorText_ = errorStream_.str();
+ goto error;
+ }
+ buffersAllocated = true;
+ stream_.state = STREAM_STOPPED;
+
+ // Set flags for buffer conversion.
+ stream_.doConvertBuffer[mode] = false;
+ if ( stream_.userFormat != stream_.deviceFormat[mode] )
+ stream_.doConvertBuffer[mode] = true;
+ if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] &&
+ stream_.nUserChannels[mode] > 1 )
+ stream_.doConvertBuffer[mode] = true;
+
+ // Allocate necessary internal buffers
+ unsigned long bufferBytes;
+ bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat );
+ stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 );
+ if ( stream_.userBuffer[mode] == NULL ) {
+ errorText_ = "RtApiAsio::probeDeviceOpen: error allocating user buffer memory.";
+ goto error;
+ }
+
+ if ( stream_.doConvertBuffer[mode] ) {
+
+ bool makeBuffer = true;
+ bufferBytes = stream_.nDeviceChannels[mode] * formatBytes( stream_.deviceFormat[mode] );
+ if ( isDuplexInput && stream_.deviceBuffer ) {
+ unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] );
+ if ( bufferBytes <= bytesOut ) makeBuffer = false;
+ }
+
+ if ( makeBuffer ) {
+ bufferBytes *= *bufferSize;
+ if ( stream_.deviceBuffer ) free( stream_.deviceBuffer );
+ stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 );
+ if ( stream_.deviceBuffer == NULL ) {
+ errorText_ = "RtApiAsio::probeDeviceOpen: error allocating device buffer memory.";
+ goto error;
+ }
+ }
+ }
+
+ // Determine device latencies
+ long inputLatency, outputLatency;
+ result = ASIOGetLatencies( &inputLatency, &outputLatency );
+ if ( result != ASE_OK ) {
+ errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error (" << getAsioErrorString( result ) << ") getting latency.";
+ errorText_ = errorStream_.str();
+ error( RTAUDIO_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 );
+
+ streamOpen = true;
+ 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();
+
+ ASIOExit();
+ 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( RTAUDIO_WARNING );
+ return;
+ }
+
+ if ( stream_.state == STREAM_RUNNING ) {
+ stream_.state = STREAM_STOPPED;
+ ASIOStop();
+ }
+
+ stream_.state = STREAM_CLOSED;
+ CallbackInfo *info = (CallbackInfo *) &stream_.callbackInfo;
+ if ( info->deviceDisconnected ) {
+ // This could be either a disconnect or a sample rate change.
+ errorText_ = "RtApiAsio: the streaming device was disconnected or the sample rate changed, closing stream!";
+ error( RTAUDIO_DEVICE_DISCONNECT );
+ }
+
+ ASIODisposeBuffers();
+ ASIOExit();
+ 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;
+ }
+
+ clearStreamInfo();
+ streamOpen = false;
+ //stream_.mode = UNINITIALIZED;
+ //stream_.state = STREAM_CLOSED;
+}
+
+RtAudioErrorType RtApiAsio :: startStream()
+{
+ if ( stream_.state != STREAM_STOPPED ) {
+ if ( stream_.state == STREAM_RUNNING )
+ errorText_ = "RtApiAsio::startStream(): the stream is already running!";
+ else if ( stream_.state == STREAM_STOPPING || stream_.state == STREAM_CLOSED )
+ errorText_ = "RtApiAsio::startStream(): the stream is stopping or closed!";
+ return error( RTAUDIO_WARNING );
+ }
+
+ /*
+ #if defined( HAVE_GETTIMEOFDAY )
+ gettimeofday( &stream_.lastTickTimestamp, NULL );
+ #endif
+ */
+
+ 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:
+ if ( result == ASE_OK ) return RTAUDIO_NO_ERROR;
+ return error( RTAUDIO_SYSTEM_ERROR );
+}
+
+RtAudioErrorType RtApiAsio :: stopStream()
+{
+ if ( stream_.state != STREAM_RUNNING && stream_.state != STREAM_STOPPING ) {
+ if ( stream_.state == STREAM_STOPPED )
+ errorText_ = "RtApiAsio::stopStream(): the stream is already stopped!";
+ else if ( stream_.state == STREAM_CLOSED )
+ errorText_ = "RtApiAsio::stopStream(): the stream is closed!";
+ return error( RTAUDIO_WARNING );
+ }
+
+ 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 RTAUDIO_NO_ERROR;
+ return error( RTAUDIO_SYSTEM_ERROR );
+}
+
+RtAudioErrorType RtApiAsio :: abortStream()
+{
+ if ( stream_.state != STREAM_RUNNING ) {
+ if ( stream_.state == STREAM_STOPPED )
+ errorText_ = "RtApiAsio::abortStream(): the stream is already stopped!";
+ else if ( stream_.state == STREAM_STOPPING || stream_.state == STREAM_CLOSED )
+ errorText_ = "RtApiAsio::abortStream(): the stream is stopping or closed!";
+ return error( RTAUDIO_WARNING );
+ }
+
+ // 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();
+ return RTAUDIO_NO_ERROR;
+}
+
+// This function will be called by a spawned thread when: 1. The user
+// callback function signals that the stream should be stopped or
+// aborted; or 2. When a signal is received indicating that the device
+// sample rate has changed or it has been disconnected. It is
+// necessary to handle it this way because the callbackEvent() or
+// signaling function must return before the ASIOStop() function will
+// return (or the driver can be removed).
+static unsigned __stdcall asioStopStream( void *ptr )
+{
+ CallbackInfo *info = (CallbackInfo *) ptr;
+ RtApiAsio *object = (RtApiAsio *) info->object;
+
+ if ( info->deviceDisconnected == false )
+ object->stopStream(); // drain the stream
+ else
+ object->closeStream(); // disconnect or sample rate change ... close the stream
+
+ _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( RTAUDIO_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;
+ if ( object->getStreamSampleRate() != sRate ) {
+ asioCallbackInfo->deviceDisconnected = true; // flag for either rate change or disconnect
+ unsigned threadId;
+ asioCallbackInfo->thread = _beginthreadex( NULL, 0, &asioStopStream,
+ asioCallbackInfo, 0, &threadId );
+ }
+}
+
+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:
+ // This message is received when a device is disconnected (and
+ // perhaps when the sample rate changes). It indicates that the
+ // driver should be reset, which is accomplished by calling
+ // ASIOStop(), ASIODisposeBuffers() and removing the driver. But
+ // since this message comes from the driver, we need to let this
+ // function return before attempting to close the stream and
+ // remove the driver. Thus, we invoke a thread to initiate the
+ // stream closing.
+ asioCallbackInfo->deviceDisconnected = true; // flag for either rate change or disconnect
+ unsigned threadId;
+ asioCallbackInfo->thread = _beginthreadex( NULL, 0, &asioStopStream,
+ asioCallbackInfo, 0, &threadId );
+ //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
+// Updates for new device selection scheme by Gary Scavone, January 2022
+// - Introduces support for the Windows WASAPI API
+// - Aims to deliver bit streams to and from hardware at the lowest possible latency, via the absolute minimum buffer sizes required
+// - Provides flexible stream configuration to an otherwise strict and inflexible WASAPI interface
+// - Includes automatic internal conversion of sample rate and buffer size between hardware and the user
+
+#ifndef INITGUID
+ #define INITGUID
+#endif
+
+#include <mfapi.h>
+#include <mferror.h>
+#include <mfplay.h>
+#include <mftransform.h>
+#include <wmcodecdsp.h>
+
+#include <audioclient.h>
+#include <avrt.h>
+#include <mmdeviceapi.h>
+#include <functiondiscoverykeys_devpkey.h>
+
+#ifndef MF_E_TRANSFORM_NEED_MORE_INPUT
+ #define MF_E_TRANSFORM_NEED_MORE_INPUT _HRESULT_TYPEDEF_(0xc00d6d72)
+#endif
+
+#ifndef MFSTARTUP_NOSOCKET
+ #define MFSTARTUP_NOSOCKET 0x1
+#endif
+
+#ifdef _MSC_VER
+ #pragma comment( lib, "ksuser" )
+ #pragma comment( lib, "mfplat.lib" )
+ #pragma comment( lib, "mfuuid.lib" )
+ #pragma comment( lib, "wmcodecdspuuid" )
+#endif
+
+//=============================================================================
+
+#define SAFE_RELEASE( objectPtr )\
+if ( objectPtr )\
+{\
+ objectPtr->Release();\
+ objectPtr = NULL;\
+}
+
+typedef HANDLE ( __stdcall *TAvSetMmThreadCharacteristicsPtr )( LPCWSTR TaskName, LPDWORD TaskIndex );
+
+#ifndef __IAudioClient3_INTERFACE_DEFINED__
+MIDL_INTERFACE( "00000000-0000-0000-0000-000000000000" ) IAudioClient3
+{
+ virtual HRESULT GetSharedModeEnginePeriod( WAVEFORMATEX*, UINT32*, UINT32*, UINT32*, UINT32* ) = 0;
+ virtual HRESULT InitializeSharedAudioStream( DWORD, UINT32, WAVEFORMATEX*, LPCGUID ) = 0;
+ virtual HRESULT Release() = 0;
+};
+#ifdef __CRT_UUID_DECL
+__CRT_UUID_DECL( IAudioClient3, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0 )
+#endif
+#endif
+
+//-----------------------------------------------------------------------------
+
+// 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_;
+ }
+
+ // the "IN" index CAN BEGIN at the "OUT" index
+ // the "IN" index CANNOT END at the "OUT" index
+ 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_;
+ }
+
+ // the "OUT" index CANNOT BEGIN at the "IN" index
+ // the "OUT" index CAN END at the "IN" index
+ if ( outIndex_ <= relInIndex && outIndexEnd > relInIndex ) {
+ return false; // not enough space between "out" index and "in" index
+ }
+
+ // copy buffer from internal to external
+ int fromZeroSize = outIndex_ + bufferSize - bufferSize_;
+ fromZeroSize = fromZeroSize < 0 ? 0 : fromZeroSize;
+ int fromOutSize = bufferSize - fromZeroSize;
+
+ switch( format )
+ {
+ case RTAUDIO_SINT8:
+ memcpy( buffer, &( ( char* ) buffer_ )[outIndex_], fromOutSize * sizeof( char ) );
+ memcpy( &( ( char* ) buffer )[fromOutSize], buffer_, fromZeroSize * sizeof( char ) );
+ break;
+ case RTAUDIO_SINT16:
+ memcpy( buffer, &( ( short* ) buffer_ )[outIndex_], fromOutSize * sizeof( short ) );
+ memcpy( &( ( short* ) buffer )[fromOutSize], buffer_, fromZeroSize * sizeof( short ) );
+ break;
+ case RTAUDIO_SINT24:
+ memcpy( buffer, &( ( S24* ) buffer_ )[outIndex_], fromOutSize * sizeof( S24 ) );
+ memcpy( &( ( S24* ) buffer )[fromOutSize], buffer_, fromZeroSize * sizeof( S24 ) );
+ break;
+ case RTAUDIO_SINT32:
+ memcpy( buffer, &( ( int* ) buffer_ )[outIndex_], fromOutSize * sizeof( int ) );
+ memcpy( &( ( int* ) buffer )[fromOutSize], buffer_, fromZeroSize * sizeof( int ) );
+ break;
+ case RTAUDIO_FLOAT32:
+ memcpy( buffer, &( ( float* ) buffer_ )[outIndex_], fromOutSize * sizeof( float ) );
+ memcpy( &( ( float* ) buffer )[fromOutSize], buffer_, fromZeroSize * sizeof( float ) );
+ break;
+ case RTAUDIO_FLOAT64:
+ memcpy( buffer, &( ( double* ) buffer_ )[outIndex_], fromOutSize * sizeof( double ) );
+ memcpy( &( ( double* ) buffer )[fromOutSize], buffer_, fromZeroSize * sizeof( double ) );
+ break;
+ }
+
+ // update "out" index
+ outIndex_ += bufferSize;
+ outIndex_ %= bufferSize_;
+
+ return true;
+ }
+
+private:
+ char* buffer_;
+ unsigned int bufferSize_;
+ unsigned int inIndex_;
+ unsigned int outIndex_;
+};
+
+//-----------------------------------------------------------------------------
+
+// In order to satisfy WASAPI's buffer requirements, we need a means of converting sample rate
+// between HW and the user. The WasapiResampler class is used to perform this conversion between
+// HwIn->UserIn and UserOut->HwOut during the stream callback loop.
+class WasapiResampler
+{
+public:
+ WasapiResampler( bool isFloat, unsigned int bitsPerSample, unsigned int channelCount,
+ unsigned int inSampleRate, unsigned int outSampleRate )
+ : _bytesPerSample( bitsPerSample / 8 )
+ , _channelCount( channelCount )
+ , _sampleRatio( ( float ) outSampleRate / inSampleRate )
+ , _transformUnk( NULL )
+ , _transform( NULL )
+ , _mediaType( NULL )
+ , _inputMediaType( NULL )
+ , _outputMediaType( NULL )
+
+ #ifdef __IWMResamplerProps_FWD_DEFINED__
+ , _resamplerProps( NULL )
+ #endif
+ {
+ // 1. Initialization
+
+ MFStartup( MF_VERSION, MFSTARTUP_NOSOCKET );
+
+ // 2. Create Resampler Transform Object
+
+ CoCreateInstance( CLSID_CResamplerMediaObject, NULL, CLSCTX_INPROC_SERVER,
+ IID_IUnknown, ( void** ) &_transformUnk );
+
+ _transformUnk->QueryInterface( IID_PPV_ARGS( &_transform ) );
+
+ #ifdef __IWMResamplerProps_FWD_DEFINED__
+ _transformUnk->QueryInterface( IID_PPV_ARGS( &_resamplerProps ) );
+ _resamplerProps->SetHalfFilterLength( 60 ); // best conversion quality
+ #endif
+
+ // 3. Specify input / output format
+
+ MFCreateMediaType( &_mediaType );
+ _mediaType->SetGUID( MF_MT_MAJOR_TYPE, MFMediaType_Audio );
+ _mediaType->SetGUID( MF_MT_SUBTYPE, isFloat ? MFAudioFormat_Float : MFAudioFormat_PCM );
+ _mediaType->SetUINT32( MF_MT_AUDIO_NUM_CHANNELS, channelCount );
+ _mediaType->SetUINT32( MF_MT_AUDIO_SAMPLES_PER_SECOND, inSampleRate );
+ _mediaType->SetUINT32( MF_MT_AUDIO_BLOCK_ALIGNMENT, _bytesPerSample * channelCount );
+ _mediaType->SetUINT32( MF_MT_AUDIO_AVG_BYTES_PER_SECOND, _bytesPerSample * channelCount * inSampleRate );
+ _mediaType->SetUINT32( MF_MT_AUDIO_BITS_PER_SAMPLE, bitsPerSample );
+ _mediaType->SetUINT32( MF_MT_ALL_SAMPLES_INDEPENDENT, TRUE );
+
+ MFCreateMediaType( &_inputMediaType );
+ _mediaType->CopyAllItems( _inputMediaType );
+
+ _transform->SetInputType( 0, _inputMediaType, 0 );
+
+ MFCreateMediaType( &_outputMediaType );
+ _mediaType->CopyAllItems( _outputMediaType );
+
+ _outputMediaType->SetUINT32( MF_MT_AUDIO_SAMPLES_PER_SECOND, outSampleRate );
+ _outputMediaType->SetUINT32( MF_MT_AUDIO_AVG_BYTES_PER_SECOND, _bytesPerSample * channelCount * outSampleRate );
+
+ _transform->SetOutputType( 0, _outputMediaType, 0 );
+
+ // 4. Send stream start messages to Resampler
+
+ _transform->ProcessMessage( MFT_MESSAGE_COMMAND_FLUSH, 0 );
+ _transform->ProcessMessage( MFT_MESSAGE_NOTIFY_BEGIN_STREAMING, 0 );
+ _transform->ProcessMessage( MFT_MESSAGE_NOTIFY_START_OF_STREAM, 0 );
+ }
+
+ ~WasapiResampler()
+ {
+ // 8. Send stream stop messages to Resampler
+
+ _transform->ProcessMessage( MFT_MESSAGE_NOTIFY_END_OF_STREAM, 0 );
+ _transform->ProcessMessage( MFT_MESSAGE_NOTIFY_END_STREAMING, 0 );
+
+ // 9. Cleanup
+
+ MFShutdown();
+
+ SAFE_RELEASE( _transformUnk );
+ SAFE_RELEASE( _transform );
+ SAFE_RELEASE( _mediaType );
+ SAFE_RELEASE( _inputMediaType );
+ SAFE_RELEASE( _outputMediaType );
+
+ #ifdef __IWMResamplerProps_FWD_DEFINED__
+ SAFE_RELEASE( _resamplerProps );
+ #endif
+ }
+
+ void Convert( char* outBuffer, const char* inBuffer, unsigned int inSampleCount, unsigned int& outSampleCount, int maxOutSampleCount = -1 )
+ {
+ unsigned int inputBufferSize = _bytesPerSample * _channelCount * inSampleCount;
+ if ( _sampleRatio == 1 )
+ {
+ // no sample rate conversion required
+ memcpy( outBuffer, inBuffer, inputBufferSize );
+ outSampleCount = inSampleCount;
+ return;
+ }
+
+ unsigned int outputBufferSize = 0;
+ if ( maxOutSampleCount != -1 )
+ {
+ outputBufferSize = _bytesPerSample * _channelCount * maxOutSampleCount;
+ }
+ else
+ {
+ outputBufferSize = ( unsigned int ) ceilf( inputBufferSize * _sampleRatio ) + ( _bytesPerSample * _channelCount );
+ }
+
+ IMFMediaBuffer* rInBuffer;
+ IMFSample* rInSample;
+ BYTE* rInByteBuffer = NULL;
+
+ // 5. Create Sample object from input data
+
+ MFCreateMemoryBuffer( inputBufferSize, &rInBuffer );
+
+ rInBuffer->Lock( &rInByteBuffer, NULL, NULL );
+ memcpy( rInByteBuffer, inBuffer, inputBufferSize );
+ rInBuffer->Unlock();
+ rInByteBuffer = NULL;
+
+ rInBuffer->SetCurrentLength( inputBufferSize );
+
+ MFCreateSample( &rInSample );
+ rInSample->AddBuffer( rInBuffer );
+
+ // 6. Pass input data to Resampler
+
+ _transform->ProcessInput( 0, rInSample, 0 );
+
+ SAFE_RELEASE( rInBuffer );
+ SAFE_RELEASE( rInSample );
+
+ // 7. Perform sample rate conversion
+
+ IMFMediaBuffer* rOutBuffer = NULL;
+ BYTE* rOutByteBuffer = NULL;
+
+ MFT_OUTPUT_DATA_BUFFER rOutDataBuffer;
+ DWORD rStatus;
+ DWORD rBytes = outputBufferSize; // maximum bytes accepted per ProcessOutput
+
+ // 7.1 Create Sample object for output data
+
+ memset( &rOutDataBuffer, 0, sizeof rOutDataBuffer );
+ MFCreateSample( &( rOutDataBuffer.pSample ) );
+ MFCreateMemoryBuffer( rBytes, &rOutBuffer );
+ rOutDataBuffer.pSample->AddBuffer( rOutBuffer );
+ rOutDataBuffer.dwStreamID = 0;
+ rOutDataBuffer.dwStatus = 0;
+ rOutDataBuffer.pEvents = NULL;
+
+ // 7.2 Get output data from Resampler
+
+ if ( _transform->ProcessOutput( 0, 1, &rOutDataBuffer, &rStatus ) == MF_E_TRANSFORM_NEED_MORE_INPUT )
+ {
+ outSampleCount = 0;
+ SAFE_RELEASE( rOutBuffer );
+ SAFE_RELEASE( rOutDataBuffer.pSample );
+ return;
+ }
+
+ // 7.3 Write output data to outBuffer
+
+ SAFE_RELEASE( rOutBuffer );
+ rOutDataBuffer.pSample->ConvertToContiguousBuffer( &rOutBuffer );
+ rOutBuffer->GetCurrentLength( &rBytes );
+
+ rOutBuffer->Lock( &rOutByteBuffer, NULL, NULL );
+ memcpy( outBuffer, rOutByteBuffer, rBytes );
+ rOutBuffer->Unlock();
+ rOutByteBuffer = NULL;
+
+ outSampleCount = rBytes / _bytesPerSample / _channelCount;
+ SAFE_RELEASE( rOutBuffer );
+ SAFE_RELEASE( rOutDataBuffer.pSample );
+ }
+
+private:
+ unsigned int _bytesPerSample;
+ unsigned int _channelCount;
+ float _sampleRatio;
+
+ IUnknown* _transformUnk;
+ IMFTransform* _transform;
+ IMFMediaType* _mediaType;
+ IMFMediaType* _inputMediaType;
+ IMFMediaType* _outputMediaType;
+
+ #ifdef __IWMResamplerProps_FWD_DEFINED__
+ IWMResamplerProps* _resamplerProps;
+ #endif
+};
+
+//-----------------------------------------------------------------------------
+
+// A structure to hold various information related to the WASAPI implementation.
+struct WasapiHandle
+{
+ IAudioClient* captureAudioClient;
+ IAudioClient* renderAudioClient;
+ IAudioCaptureClient* captureClient;
+ IAudioRenderClient* renderClient;
+ HANDLE captureEvent;
+ HANDLE renderEvent;
+
+ WasapiHandle()
+ : captureAudioClient( NULL ),
+ renderAudioClient( NULL ),
+ captureClient( NULL ),
+ renderClient( NULL ),
+ captureEvent( NULL ),
+ renderEvent( NULL ) {}
+};
+
+//-----------------------------------------------------------------------------
+
+RtApiWasapi::RtApiWasapi()
+ : coInitialized_( false ), deviceEnumerator_( NULL )
+{
+ // WASAPI can run either apartment or multi-threaded
+ HRESULT hr = CoInitialize( NULL );
+ if ( !FAILED( hr ) )
+ coInitialized_ = true;
+
+ // Instantiate device enumerator
+ hr = CoCreateInstance( __uuidof( MMDeviceEnumerator ), NULL,
+ CLSCTX_ALL, __uuidof( IMMDeviceEnumerator ),
+ ( void** ) &deviceEnumerator_ );
+
+ // If this runs on an old Windows, it will fail. Ignore and proceed.
+ if ( FAILED( hr ) )
+ deviceEnumerator_ = NULL;
+}
+
+//-----------------------------------------------------------------------------
+
+RtApiWasapi::~RtApiWasapi()
+{
+ MUTEX_LOCK( &stream_.mutex );
+ if ( stream_.state != STREAM_CLOSED )
+ {
+ MUTEX_UNLOCK( &stream_.mutex );
+ closeStream();
+ MUTEX_LOCK( &stream_.mutex );
+ }
+
+ SAFE_RELEASE( deviceEnumerator_ );
+
+ // If this object previously called CoInitialize()
+ if ( coInitialized_ )
+ CoUninitialize();
+ MUTEX_UNLOCK( &stream_.mutex );
+}
+
+//-----------------------------------------------------------------------------
+
+unsigned int RtApiWasapi::getDefaultInputDevice( void )
+{
+ IMMDevice* devicePtr = NULL;
+ LPWSTR defaultId = NULL;
+ std::string id;
+
+ if ( !deviceEnumerator_ ) return 0; // invalid ID
+ errorText_.clear();
+
+ // Get the default capture device Id.
+ HRESULT hr = deviceEnumerator_->GetDefaultAudioEndpoint( eCapture, eConsole, &devicePtr );
+ if ( FAILED( hr ) ) {
+ errorText_ = "RtApiWasapi::getDefaultInputDevice: Unable to retrieve default capture device handle.";
+ goto Release;
+ }
+
+ hr = devicePtr->GetId( &defaultId );
+ if ( FAILED( hr ) ) {
+ errorText_ = "RtApiWasapi::getDefaultInputDevice: Unable to get default capture device Id.";
+ goto Release;
+ }
+ id = convertCharPointerToStdString( defaultId );
+
+ Release:
+ SAFE_RELEASE( devicePtr );
+ CoTaskMemFree( defaultId );
+
+ if ( !errorText_.empty() ) {
+ error( RTAUDIO_DRIVER_ERROR );
+ return 0;
+ }
+
+ for ( unsigned int m=0; m<deviceIds_.size(); m++ ) {
+ if ( deviceIds_[m].first == id ) {
+ if ( deviceList_[m].isDefaultInput == false ) {
+ deviceList_[m].isDefaultInput = true;
+ for ( unsigned int j=m+1; j<deviceIds_.size(); j++ ) {
+ // make sure any remaining devices are not listed as the default
+ deviceList_[j].isDefaultInput = false;
+ }
+ }
+ return deviceList_[m].ID;
+ }
+ deviceList_[m].isDefaultInput = false;
+ }
+
+ // If not found above, then do system probe of devices and try again.
+ probeDevices();
+ for ( unsigned int m=0; m<deviceIds_.size(); m++ ) {
+ if ( deviceIds_[m].first == id ) return deviceList_[m].ID;
+ }
+
+ return 0;
+}
+
+//-----------------------------------------------------------------------------
+
+unsigned int RtApiWasapi::getDefaultOutputDevice( void )
+{
+ IMMDevice* devicePtr = NULL;
+ LPWSTR defaultId = NULL;
+ std::string id;
+
+ if ( !deviceEnumerator_ ) return 0; // invalid ID
+ errorText_.clear();
+
+ // Get the default render device Id.
+ HRESULT hr = deviceEnumerator_->GetDefaultAudioEndpoint( eRender, eConsole, &devicePtr );
+ if ( FAILED( hr ) ) {
+ errorText_ = "RtApiWasapi::getDefaultOutputDevice: Unable to retrieve default render device handle.";
+ goto Release;
+ }
+
+ hr = devicePtr->GetId( &defaultId );
+ if ( FAILED( hr ) ) {
+ errorText_ = "RtApiWasapi::getDefaultOutputDevice: Unable to get default render device Id.";
+ goto Release;
+ }
+ id = convertCharPointerToStdString( defaultId );
+
+ Release:
+ SAFE_RELEASE( devicePtr );
+ CoTaskMemFree( defaultId );
+
+ if ( !errorText_.empty() ) {
+ error( RTAUDIO_DRIVER_ERROR );
+ return 0;
+ }
+
+ for ( unsigned int m=0; m<deviceIds_.size(); m++ ) {
+ if ( deviceIds_[m].first == id ) {
+ if ( deviceList_[m].isDefaultOutput == false ) {
+ deviceList_[m].isDefaultOutput = true;
+ for ( unsigned int j=m+1; j<deviceIds_.size(); j++ ) {
+ // make sure any remaining devices are not listed as the default
+ deviceList_[j].isDefaultOutput = false;
+ }
+ }
+ return deviceList_[m].ID;
+ }
+ deviceList_[m].isDefaultOutput = false;
+ }
+
+ // If not found above, then do system probe of devices and try again.
+ probeDevices();
+ for ( unsigned int m=0; m<deviceIds_.size(); m++ ) {
+ if ( deviceIds_[m].first == id ) return deviceList_[m].ID;
+ }
+
+ return 0;
+}
+
+//-----------------------------------------------------------------------------
+
+void RtApiWasapi::probeDevices( void )
+{
+ unsigned int captureDeviceCount = 0;
+ unsigned int renderDeviceCount = 0;
+
+ IMMDeviceCollection* captureDevices = NULL;
+ IMMDeviceCollection* renderDevices = NULL;
+ IMMDevice* devicePtr = NULL;
+
+ LPWSTR defaultCaptureId = NULL;
+ LPWSTR defaultRenderId = NULL;
+ std::string defaultCaptureString;
+ std::string defaultRenderString;
+
+ unsigned int nDevices;
+ bool isCaptureDevice = false;
+ std::vector< std::pair< std::string, bool> > ids;
+ LPWSTR deviceId = NULL;
+
+ if ( !deviceEnumerator_ ) return;
+ errorText_.clear();
+
+ // Count capture devices
+ HRESULT hr = deviceEnumerator_->EnumAudioEndpoints( eCapture, DEVICE_STATE_ACTIVE, &captureDevices );
+ if ( FAILED( hr ) ) {
+ errorText_ = "RtApiWasapi::probeDevices: Unable to retrieve capture device collection.";
+ goto Exit;
+ }
+
+ hr = captureDevices->GetCount( &captureDeviceCount );
+ if ( FAILED( hr ) ) {
+ errorText_ = "RtApiWasapi::probeDevices: Unable to retrieve capture device count.";
+ goto Exit;
+ }
+
+ // Count render devices
+ hr = deviceEnumerator_->EnumAudioEndpoints( eRender, DEVICE_STATE_ACTIVE, &renderDevices );
+ if ( FAILED( hr ) ) {
+ errorText_ = "RtApiWasapi::probeDevices: Unable to retrieve render device collection.";
+ goto Exit;
+ }
+
+ hr = renderDevices->GetCount( &renderDeviceCount );
+ if ( FAILED( hr ) ) {
+ errorText_ = "RtApiWasapi::probeDevices: Unable to retrieve render device count.";
+ goto Exit;
+ }
+
+ nDevices = captureDeviceCount + renderDeviceCount;
+ if ( nDevices == 0 ) {
+ errorText_ = "RtApiWasapi::probeDevices: No devices found.";
+ goto Exit;
+ }
+
+ // Get the default capture device Id.
+ hr = deviceEnumerator_->GetDefaultAudioEndpoint( eCapture, eConsole, &devicePtr );
+ if ( SUCCEEDED( hr) ) {
+ hr = devicePtr->GetId( &defaultCaptureId );
+ if ( FAILED( hr ) ) {
+ errorText_ = "RtApiWasapi::probeDevices: Unable to get default capture device Id.";
+ goto Exit;
+ }
+ defaultCaptureString = convertCharPointerToStdString( defaultCaptureId );
+ }
+
+ // Get the default render device Id.
+ SAFE_RELEASE( devicePtr );
+ hr = deviceEnumerator_->GetDefaultAudioEndpoint( eRender, eConsole, &devicePtr );
+ if ( SUCCEEDED( hr) ) {
+ hr = devicePtr->GetId( &defaultRenderId );
+ if ( FAILED( hr ) ) {
+ errorText_ = "RtApiWasapi::probeDevices: Unable to get default render device Id.";
+ goto Exit;
+ }
+ defaultRenderString = convertCharPointerToStdString( defaultRenderId );
+ }
+
+ // Collect device IDs with mode.
+ for ( unsigned int n=0; n<nDevices; n++ ) {
+ SAFE_RELEASE( devicePtr );
+ if ( n < renderDeviceCount ) {
+ hr = renderDevices->Item( n, &devicePtr );
+ if ( FAILED( hr ) ) {
+ errorText_ = "RtApiWasapi::probeDevices: Unable to retrieve render device handle.";
+ error( RTAUDIO_WARNING );
+ continue;
+ }
+ }
+ else {
+ hr = captureDevices->Item( n - renderDeviceCount, &devicePtr );
+ if ( FAILED( hr ) ) {
+ errorText_ = "RtApiWasapi::probeDevices: Unable to retrieve capture device handle.";
+ error( RTAUDIO_WARNING );
+ continue;
+ }
+ isCaptureDevice = true;
+ }
+
+ hr = devicePtr->GetId( &deviceId );
+ if ( FAILED( hr ) ) {
+ errorText_ = "RtApiWasapi::probeDevices: Unable to get device Id.";
+ error( RTAUDIO_WARNING );
+ continue;
+ }
+
+ ids.push_back( std::pair< std::string, bool>(convertCharPointerToStdString(deviceId), isCaptureDevice) );
+ CoTaskMemFree( deviceId );
+ }
+
+ // Fill or update the deviceList_ and also save a corresponding list of Ids.
+ for ( unsigned int n=0; n<ids.size(); n++ ) {
+ if ( std::find( deviceIds_.begin(), deviceIds_.end(), ids[n] ) != deviceIds_.end() ) {
+ continue; // We already have this device.
+ }
+ else { // There is a new device to probe.
+ RtAudio::DeviceInfo info;
+ std::wstring temp = std::wstring(ids[n].first.begin(), ids[n].first.end());
+ if ( probeDeviceInfo( info, (LPWSTR) temp.c_str(), ids[n].second ) == false ) continue; // ignore if probe fails
+ deviceIds_.push_back( ids[n] );
+ info.ID = currentDeviceId_++; // arbitrary internal device ID
+ deviceList_.push_back( info );
+ }
+ }
+
+ // Remove any devices left in the list that are no longer available.
+ unsigned int m;
+ for ( std::vector< std::pair< std::string, bool> >::iterator it=deviceIds_.begin(); it!=deviceIds_.end(); ) {
+ for ( m=0; m<ids.size(); m++ ) {
+ if ( ids[m] == *it ) {
+ ++it;
+ break;
+ }
+ }
+ if ( m == ids.size() ) { // not found so remove it from our two lists
+ it = deviceIds_.erase(it);
+ deviceList_.erase( deviceList_.begin() + distance(deviceIds_.begin(), it ) );
+ }
+ }
+
+ // Set the default device flags in deviceList_.
+ for ( m=0; m<deviceList_.size(); m++ ) {
+ if ( deviceIds_[m].first == defaultRenderString )
+ deviceList_[m].isDefaultOutput = true;
+ else
+ deviceList_[m].isDefaultOutput = false;
+ if ( deviceIds_[m].first == defaultCaptureString )
+ deviceList_[m].isDefaultInput = true;
+ else
+ deviceList_[m].isDefaultInput = false;
+ }
+
+ Exit:
+ // Release all references
+ SAFE_RELEASE( captureDevices );
+ SAFE_RELEASE( renderDevices );
+ SAFE_RELEASE( devicePtr );
+
+ CoTaskMemFree( defaultCaptureId );
+ CoTaskMemFree( defaultRenderId );
+
+ if ( !errorText_.empty() ) {
+ deviceList_.clear();
+ deviceIds_.clear();
+ error( RTAUDIO_DRIVER_ERROR );
+ }
+ return;
+}
+
+//-----------------------------------------------------------------------------
+
+bool RtApiWasapi::probeDeviceInfo( RtAudio::DeviceInfo &info, LPWSTR deviceId, bool isCaptureDevice )
+{
+ PROPVARIANT deviceNameProp;
+ IMMDevice* devicePtr = NULL;
+ IAudioClient* audioClient = NULL;
+ IPropertyStore* devicePropStore = NULL;
+
+ WAVEFORMATEX* deviceFormat = NULL;
+ WAVEFORMATEX* closestMatchFormat = NULL;
+
+ errorText_.clear();
+ RtAudioErrorType errorType = RTAUDIO_DRIVER_ERROR;
+
+ // Get the device pointer from the device Id
+ HRESULT hr = deviceEnumerator_->GetDevice( deviceId, &devicePtr );
+ if ( FAILED( hr ) ) {
+ errorText_ = "RtApiWasapi::probeDeviceInfo: Unable to retrieve device handle.";
+ goto Exit;
+ }
+
+ // Get device name
+ hr = devicePtr->OpenPropertyStore( STGM_READ, &devicePropStore );
+ if ( FAILED( hr ) ) {
+ errorText_ = "RtApiWasapi::probeDeviceInfo: Unable to open device property store.";
+ goto Exit;
+ }
+
+ PropVariantInit( &deviceNameProp );
+
+ hr = devicePropStore->GetValue( PKEY_Device_FriendlyName, &deviceNameProp );
+ if ( FAILED( hr ) || deviceNameProp.pwszVal == nullptr ) {
+ errorText_ = "RtApiWasapi::probeDeviceInfo: Unable to retrieve device property: PKEY_Device_FriendlyName.";
+ goto Exit;
+ }
+
+ info.name = convertCharPointerToStdString( deviceNameProp.pwszVal );
+
+ // Get audio client
+ hr = devicePtr->Activate( __uuidof( IAudioClient ), CLSCTX_ALL, NULL, ( void** ) &audioClient );
+ if ( FAILED( hr ) ) {
+ errorText_ = "RtApiWasapi::probeDeviceInfo: Unable to retrieve device audio client.";
+ goto Exit;
+ }
+
+ hr = audioClient->GetMixFormat( &deviceFormat );
+ if ( FAILED( hr ) ) {
+ errorText_ = "RtApiWasapi::probeDeviceInfo: Unable to retrieve device mix format.";
+ goto Exit;
+ }
+
+ // Set channel count
+ if ( isCaptureDevice ) {
+ info.inputChannels = deviceFormat->nChannels;
+ info.outputChannels = 0;
+ info.duplexChannels = 0;
+ }
+ else {
+ info.inputChannels = 0;
+ info.outputChannels = deviceFormat->nChannels;
+ info.duplexChannels = 0;
+ }
+
+ // Set 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;
+
+ // Set native formats
+ 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;
+ }
+ }
+
+ Exit:
+ // Release all references
+ PropVariantClear( &deviceNameProp );
+
+ SAFE_RELEASE( devicePtr );
+ SAFE_RELEASE( audioClient );
+ SAFE_RELEASE( devicePropStore );
+
+ CoTaskMemFree( deviceFormat );
+ CoTaskMemFree( closestMatchFormat );
+
+ if ( !errorText_.empty() ) {
+ error( errorType );
+ return false;
+ }
+ return true;
+}
+
+void RtApiWasapi::closeStream( void )
+{
+ MUTEX_LOCK( &stream_.mutex );
+ if ( stream_.state == STREAM_CLOSED ) {
+ errorText_ = "RtApiWasapi::closeStream: No open stream to close.";
+ error( RTAUDIO_WARNING );
+ MUTEX_UNLOCK( &stream_.mutex );
+ return;
+ }
+
+ if ( stream_.state != STREAM_STOPPED )
+ {
+ MUTEX_UNLOCK( &stream_.mutex );
+ stopStream();
+ MUTEX_LOCK( &stream_.mutex );
+ }
+
+ // clean up stream memory
+ SAFE_RELEASE(((WasapiHandle*)stream_.apiHandle)->captureClient)
+ SAFE_RELEASE(((WasapiHandle*)stream_.apiHandle)->renderClient)
+
+ SAFE_RELEASE( ( ( WasapiHandle* ) stream_.apiHandle )->captureAudioClient )
+ SAFE_RELEASE( ( ( WasapiHandle* ) stream_.apiHandle )->renderAudioClient )
+
+ 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;
+ }
+
+ clearStreamInfo();
+ MUTEX_UNLOCK( &stream_.mutex );
+}
+
+//-----------------------------------------------------------------------------
+
+RtAudioErrorType RtApiWasapi::startStream( void )
+{
+ MUTEX_LOCK( &stream_.mutex );
+ if ( stream_.state != STREAM_STOPPED ) {
+ if ( stream_.state == STREAM_RUNNING )
+ errorText_ = "RtApiWasapi::startStream(): the stream is already running!";
+ else if ( stream_.state == STREAM_STOPPING || stream_.state == STREAM_CLOSED )
+ errorText_ = "RtApiWasapi::startStream(): the stream is stopping or closed!";
+ MUTEX_UNLOCK( &stream_.mutex );
+ return error( RTAUDIO_WARNING );
+ }
+
+ /*
+ #if defined( HAVE_GETTIMEOFDAY )
+ gettimeofday( &stream_.lastTickTimestamp, NULL );
+ #endif
+ */
+
+ // 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.";
+ MUTEX_UNLOCK( &stream_.mutex );
+ return error( RTAUDIO_THREAD_ERROR );
+ }
+ else {
+ SetThreadPriority( ( void* ) stream_.callbackInfo.thread, stream_.callbackInfo.priority );
+ ResumeThread( ( void* ) stream_.callbackInfo.thread );
+ }
+ MUTEX_UNLOCK( &stream_.mutex );
+ return RTAUDIO_NO_ERROR;
+}
+
+//-----------------------------------------------------------------------------
+
+RtAudioErrorType RtApiWasapi::stopStream( void )
+{
+ MUTEX_LOCK( &stream_.mutex );
+ if ( stream_.state != STREAM_RUNNING && stream_.state != STREAM_STOPPING ) {
+ if ( stream_.state == STREAM_STOPPED )
+ errorText_ = "RtApiWasapi::stopStream(): the stream is already stopped!";
+ else if ( stream_.state == STREAM_CLOSED )
+ errorText_ = "RtApiWasapi::stopStream(): the stream is closed!";
+ MUTEX_UNLOCK( &stream_.mutex );
+ return error( RTAUDIO_WARNING );
+ }
+
+ // inform stream thread by setting stream state to STREAM_STOPPING
+ stream_.state = STREAM_STOPPING;
+
+ WaitForSingleObject( ( void* ) stream_.callbackInfo.thread, INFINITE );
+
+ // close thread handle
+ if ( stream_.callbackInfo.thread && !CloseHandle( ( void* ) stream_.callbackInfo.thread ) ) {
+ errorText_ = "RtApiWasapi::stopStream: Unable to close callback thread.";
+ MUTEX_UNLOCK( &stream_.mutex );
+ return error( RTAUDIO_THREAD_ERROR );
+ }
+
+ stream_.callbackInfo.thread = (ThreadHandle) NULL;
+ MUTEX_UNLOCK( &stream_.mutex );
+ return RTAUDIO_NO_ERROR;
+}
+
+//-----------------------------------------------------------------------------
+
+RtAudioErrorType RtApiWasapi::abortStream( void )
+{
+ MUTEX_LOCK( &stream_.mutex );
+ if ( stream_.state != STREAM_RUNNING ) {
+ if ( stream_.state == STREAM_STOPPED )
+ errorText_ = "RtApiWasapi::abortStream(): the stream is already stopped!";
+ else if ( stream_.state == STREAM_STOPPING || stream_.state == STREAM_CLOSED )
+ errorText_ = "RtApiWasapi::abortStream(): the stream is stopping or closed!";
+ MUTEX_UNLOCK( &stream_.mutex );
+ return error( RTAUDIO_WARNING );
+ }
+
+ // inform stream thread by setting stream state to STREAM_STOPPING
+ stream_.state = STREAM_STOPPING;
+
+ WaitForSingleObject( ( void* ) stream_.callbackInfo.thread, INFINITE );
+
+ // close thread handle
+ if ( stream_.callbackInfo.thread && !CloseHandle( ( void* ) stream_.callbackInfo.thread ) ) {
+ errorText_ = "RtApiWasapi::abortStream: Unable to close callback thread.";
+ MUTEX_UNLOCK( &stream_.mutex );
+ return error( RTAUDIO_THREAD_ERROR );
+ }
+
+ stream_.callbackInfo.thread = (ThreadHandle) NULL;
+ MUTEX_UNLOCK( &stream_.mutex );
+ return RTAUDIO_NO_ERROR;
+}
+
+//-----------------------------------------------------------------------------
+
+bool RtApiWasapi::probeDeviceOpen( unsigned int deviceId, StreamMode mode, unsigned int channels,
+ unsigned int firstChannel, unsigned int sampleRate,
+ RtAudioFormat format, unsigned int* bufferSize,
+ RtAudio::StreamOptions* options )
+{
+ MUTEX_LOCK( &stream_.mutex );
+ bool methodResult = FAILURE;
+ IMMDevice* devicePtr = NULL;
+ WAVEFORMATEX* deviceFormat = NULL;
+ unsigned int bufferBytes;
+ stream_.state = STREAM_STOPPED;
+ bool isInput = false;
+ std::string id;
+
+ unsigned int deviceIdx;
+ for ( deviceIdx=0; deviceIdx<deviceList_.size(); deviceIdx++ ) {
+ if ( deviceList_[deviceIdx].ID == deviceId ) {
+ id = deviceIds_[deviceIdx].first;
+ if ( deviceIds_[deviceIdx].second ) isInput = true;
+ break;
+ }
+ }
+
+ errorText_.clear();
+ RtAudioErrorType errorType = RTAUDIO_INVALID_USE;
+ if ( id.empty() ) {
+ errorText_ = "RtApiWasapi::probeDeviceOpen: the device ID was not found!";
+ MUTEX_UNLOCK( &stream_.mutex );
+ return FAILURE;
+ }
+
+ if ( isInput && mode != INPUT ) {
+ errorText_ = "RtApiWasapi::probeDeviceOpen: deviceId specified does not support output mode.";
+ MUTEX_UNLOCK( &stream_.mutex );
+ return FAILURE;
+ }
+
+ // Get the device pointer from the device Id
+ errorType = RTAUDIO_DRIVER_ERROR;
+ std::wstring temp = std::wstring(id.begin(), id.end());
+ HRESULT hr = deviceEnumerator_->GetDevice( (LPWSTR)temp.c_str(), &devicePtr );
+ if ( FAILED( hr ) ) {
+ errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve device handle.";
+ MUTEX_UNLOCK( &stream_.mutex );
+ return FAILURE;
+ }
+
+ // Create API handle if not already created.
+ if ( !stream_.apiHandle )
+ stream_.apiHandle = ( void* ) new WasapiHandle();
+
+ if ( isInput ) {
+ IAudioClient*& captureAudioClient = ( ( WasapiHandle* ) stream_.apiHandle )->captureAudioClient;
+
+ hr = devicePtr->Activate( __uuidof( IAudioClient ), CLSCTX_ALL,
+ NULL, ( void** ) &captureAudioClient );
+ if ( FAILED( hr ) ) {
+ errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve capture device audio client.";
+ goto Exit;
+ }
+
+ hr = captureAudioClient->GetMixFormat( &deviceFormat );
+ if ( FAILED( hr ) ) {
+ errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve capture device mix format.";
+ goto Exit;
+ }
+
+ stream_.nDeviceChannels[mode] = deviceFormat->nChannels;
+ captureAudioClient->GetStreamLatency( ( long long* ) &stream_.latency[mode] );
+ }
+
+ // If an output device and is configured for loopback (input mode)
+ if ( isInput == false && mode == INPUT ) {
+ // If renderAudioClient is not initialised, initialise it now
+ IAudioClient*& renderAudioClient = ( ( WasapiHandle* ) stream_.apiHandle )->renderAudioClient;
+ if ( !renderAudioClient ) {
+ MUTEX_UNLOCK( &stream_.mutex );
+ probeDeviceOpen( deviceId, OUTPUT, channels, firstChannel, sampleRate, format, bufferSize, options );
+ MUTEX_LOCK( &stream_.mutex );
+ }
+
+ // Retrieve captureAudioClient from our stream handle.
+ IAudioClient*& captureAudioClient = ( ( WasapiHandle* ) stream_.apiHandle )->captureAudioClient;
+
+ hr = devicePtr->Activate( __uuidof( IAudioClient ), CLSCTX_ALL,
+ NULL, ( void** ) &captureAudioClient );
+ if ( FAILED( hr ) ) {
+ errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve render device audio client.";
+ goto Exit;
+ }
+
+ hr = captureAudioClient->GetMixFormat( &deviceFormat );
+ if ( FAILED( hr ) ) {
+ errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve render device mix format.";
+ goto Exit;
+ }
+
+ stream_.nDeviceChannels[mode] = deviceFormat->nChannels;
+ captureAudioClient->GetStreamLatency( ( long long* ) &stream_.latency[mode] );
+ }
+
+ // If output device and is configured for output.
+ if ( isInput == false && mode == OUTPUT ) {
+ // If renderAudioClient is already initialised, don't initialise it again
+ IAudioClient*& renderAudioClient = ( ( WasapiHandle* ) stream_.apiHandle )->renderAudioClient;
+ if ( renderAudioClient ) {
+ methodResult = SUCCESS;
+ goto Exit;
+ }
+
+ hr = devicePtr->Activate( __uuidof( IAudioClient ), CLSCTX_ALL,
+ NULL, ( void** ) &renderAudioClient );
+ if ( FAILED( hr ) ) {
+ errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve render device audio client.";
+ goto Exit;
+ }
+
+ hr = renderAudioClient->GetMixFormat( &deviceFormat );
+ if ( FAILED( hr ) ) {
+ errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve render device mix format.";
+ goto Exit;
+ }
+
+ stream_.nDeviceChannels[mode] = deviceFormat->nChannels;
+ renderAudioClient->GetStreamLatency( ( long long* ) &stream_.latency[mode] );
+ }
+
+ // Fill stream data
+ if ( ( stream_.mode == OUTPUT && mode == INPUT ) ||
+ ( stream_.mode == INPUT && mode == OUTPUT ) ) {
+ stream_.mode = DUPLEX;
+ }
+ else {
+ stream_.mode = mode;
+ }
+
+ stream_.deviceId[mode] = deviceId;
+ 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] = deviceList_[deviceIdx].nativeFormats;
+
+ if ( options && options->flags & RTAUDIO_NONINTERLEAVED )
+ stream_.userInterleaved = false;
+ else
+ stream_.userInterleaved = true;
+ stream_.deviceInterleaved[mode] = true;
+
+ // Set flags for buffer conversion.
+ stream_.doConvertBuffer[mode] = false;
+ if ( stream_.userFormat != stream_.deviceFormat[mode] ||
+ stream_.nUserChannels[0] != stream_.nDeviceChannels[0] ||
+ stream_.nUserChannels[1] != stream_.nDeviceChannels[1] )
+ stream_.doConvertBuffer[mode] = true;
+ else if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] &&
+ stream_.nUserChannels[mode] > 1 )
+ stream_.doConvertBuffer[mode] = true;
+
+ if ( stream_.doConvertBuffer[mode] )
+ setConvertInfo( mode, firstChannel );
+
+ // 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 = RTAUDIO_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( devicePtr );
+ CoTaskMemFree( deviceFormat );
+
+ // if method failed, close the stream
+ if ( methodResult == FAILURE )
+ {
+ MUTEX_UNLOCK( &stream_.mutex );
+ closeStream();
+ MUTEX_LOCK( &stream_.mutex );
+ }
+
+ if ( !errorText_.empty() )
+ error( errorType );
+
+ MUTEX_UNLOCK( &stream_.mutex );
+ return methodResult;
+}
+
+//=============================================================================
+
+DWORD WINAPI RtApiWasapi::runWasapiThread( void* wasapiPtr )
+{
+ if ( wasapiPtr )
+ ( ( RtApiWasapi* ) wasapiPtr )->wasapiThread();
+
+ return 0;
+}
+
+DWORD WINAPI RtApiWasapi::stopWasapiThread( void* wasapiPtr )
+{
+ if ( wasapiPtr )
+ ( ( RtApiWasapi* ) wasapiPtr )->stopStream();
+
+ return 0;
+}
+
+DWORD WINAPI RtApiWasapi::abortWasapiThread( void* wasapiPtr )
+{
+ if ( wasapiPtr )
+ ( ( RtApiWasapi* ) wasapiPtr )->abortStream();
+
+ return 0;
+}
+
+//-----------------------------------------------------------------------------
+
+void RtApiWasapi::wasapiThread()
+{
+ // as this is a new thread, we must CoInitialize it
+ CoInitialize( NULL );
+
+ HRESULT hr;
+
+ IAudioClient* captureAudioClient = ( ( WasapiHandle* ) stream_.apiHandle )->captureAudioClient;
+ IAudioClient* renderAudioClient = ( ( WasapiHandle* ) stream_.apiHandle )->renderAudioClient;
+ IAudioCaptureClient* captureClient = ( ( WasapiHandle* ) stream_.apiHandle )->captureClient;
+ IAudioRenderClient* renderClient = ( ( WasapiHandle* ) stream_.apiHandle )->renderClient;
+ HANDLE captureEvent = ( ( WasapiHandle* ) stream_.apiHandle )->captureEvent;
+ HANDLE renderEvent = ( ( WasapiHandle* ) stream_.apiHandle )->renderEvent;
+
+ WAVEFORMATEX* captureFormat = NULL;
+ WAVEFORMATEX* renderFormat = NULL;
+ float captureSrRatio = 0.0f;
+ float renderSrRatio = 0.0f;
+ WasapiBuffer captureBuffer;
+ WasapiBuffer renderBuffer;
+ WasapiResampler* captureResampler = NULL;
+ WasapiResampler* renderResampler = NULL;
+
+ // declare local stream variables
+ RtAudioCallback callback = ( RtAudioCallback ) stream_.callbackInfo.callback;
+ BYTE* streamBuffer = NULL;
+ DWORD captureFlags = 0;
+ unsigned int bufferFrameCount = 0;
+ unsigned int numFramesPadding = 0;
+ unsigned int convBufferSize = 0;
+ bool loopbackEnabled = stream_.deviceId[INPUT] == stream_.deviceId[OUTPUT];
+ bool callbackPushed = true;
+ bool callbackPulled = false;
+ bool callbackStopped = false;
+ int callbackResult = 0;
+
+ // convBuffer is used to store converted buffers between WASAPI and the user
+ char* convBuffer = NULL;
+ unsigned int convBuffSize = 0;
+ unsigned int deviceBuffSize = 0;
+
+ std::string errorText;
+ RtAudioErrorType errorType = RTAUDIO_DRIVER_ERROR;
+
+ // Attempt to assign "Pro Audio" characteristic to thread
+ HMODULE AvrtDll = LoadLibraryW( L"AVRT.dll" );
+ if ( AvrtDll ) {
+ DWORD taskIndex = 0;
+ TAvSetMmThreadCharacteristicsPtr AvSetMmThreadCharacteristicsPtr =
+ ( TAvSetMmThreadCharacteristicsPtr ) (void(*)()) GetProcAddress( AvrtDll, "AvSetMmThreadCharacteristicsW" );
+ AvSetMmThreadCharacteristicsPtr( L"Pro Audio", &taskIndex );
+ FreeLibrary( AvrtDll );
+ }
+
+ // start capture stream if applicable
+ if ( captureAudioClient ) {
+ hr = captureAudioClient->GetMixFormat( &captureFormat );
+ if ( FAILED( hr ) ) {
+ errorText = "RtApiWasapi::wasapiThread: Unable to retrieve device mix format.";
+ goto Exit;
+ }
+
+ // init captureResampler
+ captureResampler = new WasapiResampler( stream_.deviceFormat[INPUT] == RTAUDIO_FLOAT32 || stream_.deviceFormat[INPUT] == RTAUDIO_FLOAT64,
+ formatBytes( stream_.deviceFormat[INPUT] ) * 8, stream_.nDeviceChannels[INPUT],
+ captureFormat->nSamplesPerSec, stream_.sampleRate );
+
+ captureSrRatio = ( ( float ) captureFormat->nSamplesPerSec / stream_.sampleRate );
+
+ if ( !captureClient ) {
+ IAudioClient3* captureAudioClient3 = nullptr;
+ captureAudioClient->QueryInterface( __uuidof( IAudioClient3 ), ( void** ) &captureAudioClient3 );
+ if ( captureAudioClient3 && !loopbackEnabled )
+ {
+ UINT32 Ignore;
+ UINT32 MinPeriodInFrames;
+ hr = captureAudioClient3->GetSharedModeEnginePeriod( captureFormat,
+ &Ignore,
+ &Ignore,
+ &MinPeriodInFrames,
+ &Ignore );
+ if ( FAILED( hr ) ) {
+ errorText = "RtApiWasapi::wasapiThread: Unable to initialize capture audio client.";
+ goto Exit;
+ }
+
+ hr = captureAudioClient3->InitializeSharedAudioStream( AUDCLNT_STREAMFLAGS_EVENTCALLBACK,
+ MinPeriodInFrames,
+ captureFormat,
+ NULL );
+ SAFE_RELEASE(captureAudioClient3);
+ }
+ else
+ {
+ hr = captureAudioClient->Initialize( AUDCLNT_SHAREMODE_SHARED,
+ loopbackEnabled ? AUDCLNT_STREAMFLAGS_LOOPBACK : AUDCLNT_STREAMFLAGS_EVENTCALLBACK,
+ 0,
+ 0,
+ captureFormat,
+ NULL );
+ }
+
+ if ( FAILED( hr ) ) {
+ errorText = "RtApiWasapi::wasapiThread: Unable to initialize capture audio client.";
+ goto Exit;
+ }
+
+ hr = captureAudioClient->GetService( __uuidof( IAudioCaptureClient ),
+ ( void** ) &captureClient );
+ if ( FAILED( hr ) ) {
+ errorText = "RtApiWasapi::wasapiThread: Unable to retrieve capture client handle.";
+ goto Exit;
+ }
+
+ // don't configure captureEvent if in loopback mode
+ if ( !loopbackEnabled )
+ {
+ // configure captureEvent to trigger on every available capture buffer
+ captureEvent = CreateEvent( NULL, FALSE, FALSE, NULL );
+ if ( !captureEvent ) {
+ errorType = RTAUDIO_SYSTEM_ERROR;
+ errorText = "RtApiWasapi::wasapiThread: Unable to create capture event.";
+ goto Exit;
+ }
+
+ hr = captureAudioClient->SetEventHandle( captureEvent );
+ if ( FAILED( hr ) ) {
+ errorText = "RtApiWasapi::wasapiThread: Unable to set capture event handle.";
+ goto Exit;
+ }
+
+ ( ( WasapiHandle* ) stream_.apiHandle )->captureEvent = captureEvent;
+ }
+
+ ( ( WasapiHandle* ) stream_.apiHandle )->captureClient = captureClient;
+
+ // 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;
+ }
+ }
+
+ unsigned int inBufferSize = 0;
+ hr = captureAudioClient->GetBufferSize( &inBufferSize );
+ if ( FAILED( hr ) ) {
+ errorText = "RtApiWasapi::wasapiThread: Unable to get capture buffer size.";
+ goto Exit;
+ }
+
+ // scale outBufferSize according to stream->user sample rate ratio
+ unsigned int outBufferSize = ( unsigned int ) ceilf( stream_.bufferSize * captureSrRatio ) * stream_.nDeviceChannels[INPUT];
+ inBufferSize *= stream_.nDeviceChannels[INPUT];
+
+ // set captureBuffer size
+ captureBuffer.setBufferSize( inBufferSize + outBufferSize, formatBytes( stream_.deviceFormat[INPUT] ) );
+ }
+
+ // start render stream if applicable
+ if ( renderAudioClient ) {
+ hr = renderAudioClient->GetMixFormat( &renderFormat );
+ if ( FAILED( hr ) ) {
+ errorText = "RtApiWasapi::wasapiThread: Unable to retrieve device mix format.";
+ goto Exit;
+ }
+
+ // init renderResampler
+ renderResampler = new WasapiResampler( stream_.deviceFormat[OUTPUT] == RTAUDIO_FLOAT32 || stream_.deviceFormat[OUTPUT] == RTAUDIO_FLOAT64,
+ formatBytes( stream_.deviceFormat[OUTPUT] ) * 8, stream_.nDeviceChannels[OUTPUT],
+ stream_.sampleRate, renderFormat->nSamplesPerSec );
+
+ renderSrRatio = ( ( float ) renderFormat->nSamplesPerSec / stream_.sampleRate );
+
+ if ( !renderClient ) {
+ IAudioClient3* renderAudioClient3 = nullptr;
+ renderAudioClient->QueryInterface( __uuidof( IAudioClient3 ), ( void** ) &renderAudioClient3 );
+ if ( renderAudioClient3 )
+ {
+ UINT32 Ignore;
+ UINT32 MinPeriodInFrames;
+ hr = renderAudioClient3->GetSharedModeEnginePeriod( renderFormat,
+ &Ignore,
+ &Ignore,
+ &MinPeriodInFrames,
+ &Ignore );
+ if ( FAILED( hr ) ) {
+ errorText = "RtApiWasapi::wasapiThread: Unable to initialize render audio client.";
+ goto Exit;
+ }
+
+ hr = renderAudioClient3->InitializeSharedAudioStream( AUDCLNT_STREAMFLAGS_EVENTCALLBACK,
+ MinPeriodInFrames,
+ renderFormat,
+ NULL );
+ SAFE_RELEASE(renderAudioClient3);
+ }
+ else
+ {
+ hr = renderAudioClient->Initialize( AUDCLNT_SHAREMODE_SHARED,
+ AUDCLNT_STREAMFLAGS_EVENTCALLBACK,
+ 0,
+ 0,
+ renderFormat,
+ NULL );
+ }
+
+ if ( FAILED( hr ) ) {
+ errorText = "RtApiWasapi::wasapiThread: Unable to initialize render audio client.";
+ goto Exit;
+ }
+
+ hr = renderAudioClient->GetService( __uuidof( IAudioRenderClient ),
+ ( void** ) &renderClient );
+ if ( FAILED( hr ) ) {
+ errorText = "RtApiWasapi::wasapiThread: Unable to retrieve render client handle.";
+ goto Exit;
+ }
+
+ // configure renderEvent to trigger on every available render buffer
+ renderEvent = CreateEvent( NULL, FALSE, FALSE, NULL );
+ if ( !renderEvent ) {
+ errorType = RTAUDIO_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;
+
+ // 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;
+ }
+ }
+
+ unsigned int outBufferSize = 0;
+ hr = renderAudioClient->GetBufferSize( &outBufferSize );
+ if ( FAILED( hr ) ) {
+ errorText = "RtApiWasapi::wasapiThread: Unable to get render buffer size.";
+ goto Exit;
+ }
+
+ // scale inBufferSize according to user->stream sample rate ratio
+ unsigned int inBufferSize = ( unsigned int ) ceilf( stream_.bufferSize * renderSrRatio ) * stream_.nDeviceChannels[OUTPUT];
+ outBufferSize *= stream_.nDeviceChannels[OUTPUT];
+
+ // set renderBuffer size
+ renderBuffer.setBufferSize( inBufferSize + outBufferSize, formatBytes( stream_.deviceFormat[OUTPUT] ) );
+ }
+
+ // malloc buffer memory
+ if ( stream_.mode == INPUT )
+ {
+ using namespace std; // for ceilf
+ convBuffSize = ( unsigned int ) ( ceilf( stream_.bufferSize * captureSrRatio ) ) * stream_.nDeviceChannels[INPUT] * formatBytes( stream_.deviceFormat[INPUT] );
+ deviceBuffSize = stream_.bufferSize * stream_.nDeviceChannels[INPUT] * formatBytes( stream_.deviceFormat[INPUT] );
+ }
+ else if ( stream_.mode == OUTPUT )
+ {
+ convBuffSize = ( unsigned int ) ( ceilf( stream_.bufferSize * renderSrRatio ) ) * stream_.nDeviceChannels[OUTPUT] * formatBytes( stream_.deviceFormat[OUTPUT] );
+ deviceBuffSize = stream_.bufferSize * stream_.nDeviceChannels[OUTPUT] * formatBytes( stream_.deviceFormat[OUTPUT] );
+ }
+ else if ( stream_.mode == DUPLEX )
+ {
+ convBuffSize = std::max( ( unsigned int ) ( ceilf( stream_.bufferSize * captureSrRatio ) ) * stream_.nDeviceChannels[INPUT] * formatBytes( stream_.deviceFormat[INPUT] ),
+ ( unsigned int ) ( ceilf( stream_.bufferSize * renderSrRatio ) ) * stream_.nDeviceChannels[OUTPUT] * formatBytes( stream_.deviceFormat[OUTPUT] ) );
+ deviceBuffSize = std::max( stream_.bufferSize * stream_.nDeviceChannels[INPUT] * formatBytes( stream_.deviceFormat[INPUT] ),
+ stream_.bufferSize * stream_.nDeviceChannels[OUTPUT] * formatBytes( stream_.deviceFormat[OUTPUT] ) );
+ }
+
+ convBuffSize *= 2; // allow overflow for *SrRatio remainders
+ convBuffer = ( char* ) calloc( convBuffSize, 1 );
+ stream_.deviceBuffer = ( char* ) calloc( deviceBuffSize, 1 );
+ if ( !convBuffer || !stream_.deviceBuffer ) {
+ errorType = RTAUDIO_MEMORY_ERROR;
+ errorText = "RtApiWasapi::wasapiThread: Error allocating device buffer memory.";
+ goto Exit;
+ }
+
+ // stream process loop
+ while ( stream_.state != STREAM_STOPPING ) {
+ if ( !callbackPulled ) {
+ // Callback Input
+ // ==============
+ // 1. Pull callback buffer from inputBuffer
+ // 2. If 1. was successful: Convert callback buffer to user sample rate and channel count
+ // Convert callback buffer to user format
+
+ if ( captureAudioClient )
+ {
+ int samplesToPull = ( unsigned int ) floorf( stream_.bufferSize * captureSrRatio );
+
+ convBufferSize = 0;
+ while ( convBufferSize < stream_.bufferSize )
+ {
+ // Pull callback buffer from inputBuffer
+ callbackPulled = captureBuffer.pullBuffer( convBuffer,
+ samplesToPull * stream_.nDeviceChannels[INPUT],
+ stream_.deviceFormat[INPUT] );
+
+ if ( !callbackPulled )
+ {
+ break;
+ }
+
+ // Convert callback buffer to user sample rate
+ unsigned int deviceBufferOffset = convBufferSize * stream_.nDeviceChannels[INPUT] * formatBytes( stream_.deviceFormat[INPUT] );
+ unsigned int convSamples = 0;
+
+ captureResampler->Convert( stream_.deviceBuffer + deviceBufferOffset,
+ convBuffer,
+ samplesToPull,
+ convSamples,
+ convBufferSize == 0 ? -1 : stream_.bufferSize - convBufferSize );
+
+ convBufferSize += convSamples;
+ samplesToPull = 1; // now pull one sample at a time until we have stream_.bufferSize samples
+ }
+
+ if ( callbackPulled )
+ {
+ if ( stream_.doConvertBuffer[INPUT] ) {
+ // Convert callback buffer to user format
+ convertBuffer( stream_.userBuffer[INPUT],
+ stream_.deviceBuffer,
+ stream_.convertInfo[INPUT] );
+ }
+ else {
+ // no further conversion, simple copy deviceBuffer to userBuffer
+ memcpy( stream_.userBuffer[INPUT],
+ stream_.deviceBuffer,
+ stream_.bufferSize * stream_.nUserChannels[INPUT] * formatBytes( stream_.userFormat ) );
+ }
+ }
+ }
+ else {
+ // if there is no capture stream, set callbackPulled flag
+ callbackPulled = true;
+ }
+
+ // Execute Callback
+ // ================
+ // 1. Execute user callback method
+ // 2. Handle return value from callback
+
+ // if callback has not requested the stream to stop
+ if ( callbackPulled && !callbackStopped ) {
+ // Execute user callback method
+ callbackResult = callback( stream_.userBuffer[OUTPUT],
+ stream_.userBuffer[INPUT],
+ stream_.bufferSize,
+ getStreamTime(),
+ captureFlags & AUDCLNT_BUFFERFLAGS_DATA_DISCONTINUITY ? RTAUDIO_INPUT_OVERFLOW : 0,
+ stream_.callbackInfo.userData );
+
+ // tick stream time
+ RtApi::tickStreamTime();
+
+ // 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 = RTAUDIO_THREAD_ERROR;
+ errorText = "RtApiWasapi::wasapiThread: Unable to instantiate stream stop thread.";
+ goto Exit;
+ }
+ else if ( !CloseHandle( threadHandle ) ) {
+ errorType = RTAUDIO_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 = RTAUDIO_THREAD_ERROR;
+ errorText = "RtApiWasapi::wasapiThread: Unable to instantiate stream abort thread.";
+ goto Exit;
+ }
+ else if ( !CloseHandle( threadHandle ) ) {
+ errorType = RTAUDIO_THREAD_ERROR;
+ errorText = "RtApiWasapi::wasapiThread: Unable to close stream abort thread handle.";
+ goto Exit;
+ }
+
+ callbackStopped = true;
+ }
+ }
+ }
+
+ // Callback Output
+ // ===============
+ // 1. Convert callback buffer to stream format
+ // 2. Convert callback buffer to stream sample rate and channel count
+ // 3. Push callback buffer into outputBuffer
+
+ if ( renderAudioClient && callbackPulled )
+ {
+ // if the last call to renderBuffer.PushBuffer() was successful
+ if ( callbackPushed || convBufferSize == 0 )
+ {
+ if ( stream_.doConvertBuffer[OUTPUT] )
+ {
+ // Convert callback buffer to stream format
+ convertBuffer( stream_.deviceBuffer,
+ stream_.userBuffer[OUTPUT],
+ stream_.convertInfo[OUTPUT] );
+
+ }
+ else {
+ // no further conversion, simple copy userBuffer to deviceBuffer
+ memcpy( stream_.deviceBuffer,
+ stream_.userBuffer[OUTPUT],
+ stream_.bufferSize * stream_.nUserChannels[OUTPUT] * formatBytes( stream_.userFormat ) );
+ }
+
+ // Convert callback buffer to stream sample rate
+ renderResampler->Convert( convBuffer,
+ stream_.deviceBuffer,
+ stream_.bufferSize,
+ convBufferSize );
+ }
+
+ // Push callback buffer into outputBuffer
+ callbackPushed = renderBuffer.pushBuffer( convBuffer,
+ convBufferSize * stream_.nDeviceChannels[OUTPUT],
+ stream_.deviceFormat[OUTPUT] );
+ }
+ else {
+ // if there is no render stream, set callbackPushed flag
+ callbackPushed = true;
+ }
+
+ // Stream Capture
+ // ==============
+ // 1. Get capture buffer from stream
+ // 2. Push capture buffer into inputBuffer
+ // 3. If 2. was successful: Release capture buffer
+
+ if ( captureAudioClient ) {
+ // if the callback input buffer was not pulled from captureBuffer, wait for next capture event
+ if ( !callbackPulled ) {
+ WaitForSingleObject( loopbackEnabled ? renderEvent : captureEvent, INFINITE );
+ }
+
+ // Get capture buffer from stream
+ hr = captureClient->GetBuffer( &streamBuffer,
+ &bufferFrameCount,
+ &captureFlags, NULL, NULL );
+ if ( FAILED( hr ) ) {
+ errorText = "RtApiWasapi::wasapiThread: Unable to retrieve capture buffer.";
+ goto Exit;
+ }
+
+ if ( bufferFrameCount != 0 ) {
+ // Push capture buffer into inputBuffer
+ if ( captureBuffer.pushBuffer( ( char* ) streamBuffer,
+ bufferFrameCount * stream_.nDeviceChannels[INPUT],
+ stream_.deviceFormat[INPUT] ) )
+ {
+ // Release capture buffer
+ hr = captureClient->ReleaseBuffer( bufferFrameCount );
+ if ( FAILED( hr ) ) {
+ errorText = "RtApiWasapi::wasapiThread: Unable to release capture buffer.";
+ goto Exit;
+ }
+ }
+ else
+ {
+ // Inform WASAPI that capture was unsuccessful
+ hr = captureClient->ReleaseBuffer( 0 );
+ if ( FAILED( hr ) ) {
+ errorText = "RtApiWasapi::wasapiThread: Unable to release capture buffer.";
+ goto Exit;
+ }
+ }
+ }
+ else
+ {
+ // Inform WASAPI that capture was unsuccessful
+ hr = captureClient->ReleaseBuffer( 0 );
+ if ( FAILED( hr ) ) {
+ errorText = "RtApiWasapi::wasapiThread: Unable to release capture buffer.";
+ goto Exit;
+ }
+ }
+ }
+
+ // Stream Render
+ // =============
+ // 1. Get render buffer from stream
+ // 2. Pull next buffer from outputBuffer
+ // 3. If 2. was successful: Fill render buffer with next buffer
+ // Release render buffer
+
+ if ( renderAudioClient ) {
+ // if the callback output buffer was not pushed to renderBuffer, wait for next render event
+ if ( callbackPulled && !callbackPushed ) {
+ WaitForSingleObject( renderEvent, INFINITE );
+ }
+
+ // Get render buffer from stream
+ hr = renderAudioClient->GetBufferSize( &bufferFrameCount );
+ if ( FAILED( hr ) ) {
+ errorText = "RtApiWasapi::wasapiThread: Unable to retrieve render buffer size.";
+ goto Exit;
+ }
+
+ hr = renderAudioClient->GetCurrentPadding( &numFramesPadding );
+ if ( FAILED( hr ) ) {
+ errorText = "RtApiWasapi::wasapiThread: Unable to retrieve render buffer padding.";
+ goto Exit;
+ }
+
+ bufferFrameCount -= numFramesPadding;
+
+ if ( bufferFrameCount != 0 ) {
+ hr = renderClient->GetBuffer( bufferFrameCount, &streamBuffer );
+ if ( FAILED( hr ) ) {
+ errorText = "RtApiWasapi::wasapiThread: Unable to retrieve render buffer.";
+ goto Exit;
+ }
+
+ // Pull next buffer from outputBuffer
+ // Fill render buffer with next buffer
+ if ( renderBuffer.pullBuffer( ( char* ) streamBuffer,
+ bufferFrameCount * stream_.nDeviceChannels[OUTPUT],
+ stream_.deviceFormat[OUTPUT] ) )
+ {
+ // Release render buffer
+ hr = renderClient->ReleaseBuffer( bufferFrameCount, 0 );
+ if ( FAILED( hr ) ) {
+ errorText = "RtApiWasapi::wasapiThread: Unable to release render buffer.";
+ goto Exit;
+ }
+ }
+ else
+ {
+ // Inform WASAPI that render was unsuccessful
+ hr = renderClient->ReleaseBuffer( 0, 0 );
+ if ( FAILED( hr ) ) {
+ errorText = "RtApiWasapi::wasapiThread: Unable to release render buffer.";
+ goto Exit;
+ }
+ }
+ }
+ else
+ {
+ // Inform WASAPI that render was unsuccessful
+ hr = renderClient->ReleaseBuffer( 0, 0 );
+ if ( FAILED( hr ) ) {
+ errorText = "RtApiWasapi::wasapiThread: Unable to release render buffer.";
+ goto Exit;
+ }
+ }
+ }
+
+ // if the callback buffer was pushed renderBuffer reset callbackPulled flag
+ if ( callbackPushed ) {
+ // unsetting the callbackPulled flag lets the stream know that
+ // the audio device is ready for another callback output buffer.
+ callbackPulled = false;
+ }
+
+ }
+
+Exit:
+ // clean up
+ CoTaskMemFree( captureFormat );
+ CoTaskMemFree( renderFormat );
+
+ free ( convBuffer );
+ delete renderResampler;
+ delete captureResampler;
+
+ CoUninitialize();
+
+ if ( !errorText.empty() )
+ {
+ errorText_ = errorText;
+ error( errorType );
+ }
+
+ // update stream state
+ stream_.state = STREAM_STOPPED;
+}
+
+//******************** End of __WINDOWS_WASAPI__ *********************//
+#endif
+
+
+#if defined(__WINDOWS_DS__) // Windows DirectSound API
+
+// Modified by Robin Davies, October 2005
+// - Improvements to DirectX pointer chasing.
+// - Bug fix for non-power-of-two Asio granularity used by Edirol PCR-A30.
+// - Auto-call CoInitialize for DSOUND and ASIO platforms.
+// Various revisions for RtAudio 4.0 by Gary Scavone, April 2007
+// Changed device query structure for RtAudio 4.0.7, January 2010
+
+#include <windows.h>
+#include <process.h>
+#include <mmsystem.h>
+#include <mmreg.h>
+#include <dsound.h>
+#include <assert.h>
+#include <algorithm>
+
+#if defined(__MINGW32__)
+ // missing from latest mingw winapi
+#define WAVE_FORMAT_96M08 0x00010000 /* 96 kHz, Mono, 8-bit */
+#define WAVE_FORMAT_96S08 0x00020000 /* 96 kHz, Stereo, 8-bit */
+#define WAVE_FORMAT_96M16 0x00040000 /* 96 kHz, Mono, 16-bit */
+#define WAVE_FORMAT_96S16 0x00080000 /* 96 kHz, Stereo, 16-bit */
+#endif
+
+#define MINIMUM_DEVICE_BUFFER_SIZE 32768
+
+#ifdef _MSC_VER // if Microsoft Visual C++
+#pragma comment( lib, "winmm.lib" ) // then, auto-link winmm.lib. Otherwise, it has to be added manually.
+#endif
+
+static inline DWORD dsPointerBetween( DWORD pointer, DWORD laterPointer, DWORD earlierPointer, DWORD bufferSize )
+{
+ if ( pointer > bufferSize ) pointer -= bufferSize;
+ if ( laterPointer < earlierPointer ) laterPointer += bufferSize;
+ if ( pointer < earlierPointer ) pointer += bufferSize;
+ return pointer >= earlierPointer && pointer < laterPointer;
+}
+
+// A structure to hold various information related to the DirectSound
+// API implementation.
+struct DsHandle {
+ unsigned int drainCounter; // Tracks callback counts when draining
+ bool internalDrain; // Indicates if stop is initiated from callback or not.
+ void *id[2];
+ void *buffer[2];
+ bool xrun[2];
+ UINT bufferPointer[2];
+ DWORD dsBufferSize[2];
+ DWORD dsPointerLeadTime[2]; // the number of bytes ahead of the safe pointer to lead by.
+ HANDLE condition;
+
+ DsHandle()
+ :drainCounter(0), internalDrain(false) { id[0] = 0; id[1] = 0; buffer[0] = 0; buffer[1] = 0; xrun[0] = false; xrun[1] = false; bufferPointer[0] = 0; bufferPointer[1] = 0; }
+};
+
+// Declarations for utility functions, callbacks, and structures
+// specific to the DirectSound implementation.
+static BOOL CALLBACK deviceQueryCallback( LPGUID lpguid,
+ LPCTSTR description,
+ LPCTSTR module,
+ LPVOID lpContext );
+
+static const char* getErrorString( int code );
+
+static unsigned __stdcall callbackHandler( void *ptr );
+
+struct DsDevice {
+ LPGUID id;
+ bool isInput;
+ std::string name;
+ std::string epID; // endpoint ID
+
+ DsDevice()
+ : isInput(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.
+}
+
+void RtApiDs :: probeDevices( void )
+{
+ // See list of required functionality in RtApi::probeDevices().
+
+ // Query DirectSound devices.
+ struct DsProbeData probeInfo;
+ probeInfo.isInput = false;
+ std::vector< struct DsDevice > devices;
+ probeInfo.dsDevices = &devices;
+ HRESULT result = DirectSoundEnumerate( (LPDSENUMCALLBACK) deviceQueryCallback, &probeInfo );
+ if ( FAILED( result ) ) {
+ errorStream_ << "RtApiDs::probeDevices: error (" << getErrorString( result ) << ") enumerating output devices!";
+ errorText_ = errorStream_.str();
+ error( RTAUDIO_WARNING );
+ }
+
+ // Query DirectSoundCapture devices.
+ probeInfo.isInput = true;
+ result = DirectSoundCaptureEnumerate( (LPDSENUMCALLBACK) deviceQueryCallback, &probeInfo );
+ if ( FAILED( result ) ) {
+ errorStream_ << "RtApiDs::probeDevices: error (" << getErrorString( result ) << ") enumerating input devices!";
+ errorText_ = errorStream_.str();
+ error( RTAUDIO_WARNING );
+ }
+
+ // Now fill or update our deviceList_ vector.
+ unsigned int m, n;
+ for ( n=0; n<devices.size(); n++ ) {
+ for ( m=0; m<dsDevices_.size(); m++ ) {
+ if ( ( dsDevices_[m].epID == devices[n].epID ) && ( devices[n].isInput == dsDevices_[m].isInput ) ) {
+ dsDevices_[m].id = devices[n].id; // Update the ID, since it seems to change when devices are added/removed
+ break; // We already have this device.
+ }
+ }
+ if ( m == dsDevices_.size() ) { // new device
+ RtAudio::DeviceInfo info;
+ if ( probeDeviceInfo( info, devices[n] ) == false ) continue; // ignore if probe fails
+ info.ID = currentDeviceId_++; // arbitrary internal device ID
+ deviceList_.push_back( info );
+ dsDevices_.push_back( devices[n] );
+ }
+ }
+
+ // Remove any devices left in the list that are no longer available.
+ for ( std::vector< struct DsDevice >::iterator it=dsDevices_.begin(); it!=dsDevices_.end(); ) {
+ for ( n=0; n<devices.size(); n++ ) {
+ if ( (*it).epID == devices[n].epID ) {
+ ++it;
+ break;
+ }
+ }
+ if ( n == devices.size() ) { // not found so remove it from our list
+ it = dsDevices_.erase( it );
+ deviceList_.erase( deviceList_.begin() + distance(dsDevices_.begin(), it ) );
+ }
+ }
+
+ // Determine the default devices
+ for ( n=0; n<dsDevices_.size(); n++ ) {
+ if ( dsDevices_[n].id == NULL ) { // default device
+ if ( dsDevices_[n].isInput )
+ deviceList_[n].isDefaultInput = true;
+ else
+ deviceList_[n].isDefaultOutput = true;
+ }
+ else if ( dsDevices_[n].isInput )
+ deviceList_[n].isDefaultInput = false;
+ else
+ deviceList_[n].isDefaultOutput = false;
+ }
+}
+
+bool RtApiDs :: probeDeviceInfo( RtAudio::DeviceInfo &info, DsDevice &dsDevice )
+{
+ // Devices will either be input or output devices but not both.
+ HRESULT result;
+ if ( dsDevice.isInput ) goto probeInput;
+
+ LPDIRECTSOUND output;
+ DSCAPS outCaps;
+ result = DirectSoundCreate( dsDevice.id, &output, NULL );
+ if ( FAILED( result ) ) {
+ errorStream_ << "RtApiDs::probeDeviceInfo: error (" << getErrorString( result ) << ") opening output device (" << dsDevice.name << ")!";
+ errorText_ = errorStream_.str();
+ error( RTAUDIO_WARNING );
+ goto probeInput;
+ }
+
+ outCaps.dwSize = sizeof( outCaps );
+ result = output->GetCaps( &outCaps );
+ if ( FAILED( result ) ) {
+ output->Release();
+ errorStream_ << "RtApiDs::probeDeviceInfo: error (" << getErrorString( result ) << ") getting capabilities!";
+ errorText_ = errorStream_.str();
+ error( RTAUDIO_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();
+
+ info.name = dsDevice.name;
+ return true;
+
+ probeInput:
+
+ LPDIRECTSOUNDCAPTURE input;
+ result = DirectSoundCaptureCreate( dsDevice.id, &input, NULL );
+ if ( FAILED( result ) ) {
+ errorStream_ << "RtApiDs::probeDeviceInfo: error (" << getErrorString( result ) << ") opening input device (" << dsDevice.name << ")!";
+ errorText_ = errorStream_.str();
+ error( RTAUDIO_WARNING );
+ return false;
+ }
+
+ DSCCAPS inCaps;
+ inCaps.dwSize = sizeof( inCaps );
+ result = input->GetCaps( &inCaps );
+ if ( FAILED( result ) ) {
+ input->Release();
+ errorStream_ << "RtApiDs::probeDeviceInfo: error (" << getErrorString( result ) << ") getting object capabilities (" << dsDevice.name << ")!";
+ errorText_ = errorStream_.str();
+ error( RTAUDIO_WARNING );
+ return false;
+ }
+
+ // 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 false;
+
+ // 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() );
+ for ( unsigned int i=0; i<info.sampleRates.size(); i++ ) {
+ if ( !info.preferredSampleRate || ( info.sampleRates[i] <= 48000 && info.sampleRates[i] > info.preferredSampleRate ) )
+ info.preferredSampleRate = info.sampleRates[i];
+ }
+
+ // Copy name and return.
+ info.name = dsDevice.name;
+ return true;
+}
+
+bool RtApiDs :: probeDeviceOpen( unsigned int deviceId, 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;
+ }
+
+ int deviceIdx = -1;
+ for ( unsigned int m=0; m<deviceList_.size(); m++ ) {
+ if ( deviceList_[m].ID == deviceId ) {
+ deviceIdx = m;
+ break;
+ }
+ }
+
+ if ( deviceIdx < 0 ) {
+ errorText_ = "RtApiDs::probeDeviceOpen: device ID is invalid!";
+ return FAILURE;
+ }
+
+ if ( mode == OUTPUT ) {
+ if ( dsDevices_[ deviceIdx ].isInput ) {
+ errorStream_ << "RtApiDs::probeDeviceOpen: device (" << deviceIdx << ") does not support output!";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
+ }
+ else { // mode == INPUT
+ if ( dsDevices_[ deviceIdx ].isInput == false ) {
+ errorStream_ << "RtApiDs::probeDeviceOpen: device (" << deviceIdx << ") 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_[ deviceIdx ].id, &output, NULL );
+ if ( FAILED( result ) ) {
+ errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") opening output device (" << dsDevices_[ deviceIdx ].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_[ deviceIdx ].name << ")!";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
+
+ // Check channel information.
+ if ( channels + firstChannel == 2 && !( outCaps.dwFlags & DSCAPS_PRIMARYSTEREO ) ) {
+ errorStream_ << "RtApiDs::probeDeviceInfo: the output device (" << dsDevices_[ deviceIdx ].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_[ deviceIdx ].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_[ deviceIdx ].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_[ deviceIdx ].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_[ deviceIdx ].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_[ deviceIdx ].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_[ deviceIdx ].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_[ deviceIdx ].name << ")!";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
+
+ ohandle = (void *) output;
+ bhandle = (void *) buffer;
+ }
+
+ if ( mode == INPUT ) {
+
+ LPDIRECTSOUNDCAPTURE input;
+ result = DirectSoundCaptureCreate( dsDevices_[ deviceIdx ].id, &input, NULL );
+ if ( FAILED( result ) ) {
+ errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") opening input device (" << dsDevices_[ deviceIdx ].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_[ deviceIdx ].name << ")!";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
+
+ // Check channel information.
+ if ( inCaps.dwChannels < channels + firstChannel ) {
+ errorText_ = "RtApiDs::probeDeviceInfo: 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_[ deviceIdx ].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_[ deviceIdx ].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_[ deviceIdx ].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_[ deviceIdx ].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_.deviceId[mode] = deviceIdx;
+ 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( RTAUDIO_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;
+ }
+
+ clearStreamInfo();
+ //stream_.mode = UNINITIALIZED;
+ //stream_.state = STREAM_CLOSED;
+}
+
+RtAudioErrorType RtApiDs :: startStream()
+{
+ if ( stream_.state != STREAM_STOPPED ) {
+ if ( stream_.state == STREAM_RUNNING )
+ errorText_ = "RtApiDs::startStream(): the stream is already running!";
+ else if ( stream_.state == STREAM_STOPPING || stream_.state == STREAM_CLOSED )
+ errorText_ = "RtApiDs::startStream(): the stream is stopping or closed!";
+ return error( RTAUDIO_WARNING );
+ }
+
+ /*
+ #if defined( HAVE_GETTIMEOFDAY )
+ gettimeofday( &stream_.lastTickTimestamp, NULL );
+ #endif
+ */
+
+ 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( RTAUDIO_SYSTEM_ERROR );
+ return RTAUDIO_NO_ERROR;
+}
+
+RtAudioErrorType RtApiDs :: stopStream()
+{
+ if ( stream_.state != STREAM_RUNNING && stream_.state != STREAM_STOPPING ) {
+ if ( stream_.state == STREAM_STOPPED )
+ errorText_ = "RtApiDs::stopStream(): the stream is already stopped!";
+ else if ( stream_.state == STREAM_CLOSED )
+ errorText_ = "RtApiDs::stopStream(): the stream is closed!";
+ return error( RTAUDIO_WARNING );
+ }
+
+ 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( RTAUDIO_SYSTEM_ERROR );
+ return RTAUDIO_NO_ERROR;
+}
+
+RtAudioErrorType RtApiDs :: abortStream()
+{
+ if ( stream_.state != STREAM_RUNNING ) {
+ if ( stream_.state == STREAM_STOPPED )
+ errorText_ = "RtApiDs::abortStream(): the stream is already stopped!";
+ else if ( stream_.state == STREAM_STOPPING || stream_.state == STREAM_CLOSED )
+ errorText_ = "RtApiDs::abortStream(): the stream is stopping or closed!";
+ return error( RTAUDIO_WARNING );
+ }
+
+ DsHandle *handle = (DsHandle *) stream_.apiHandle;
+ handle->drainCounter = 2;
+
+ return 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( RTAUDIO_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( RTAUDIO_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( RTAUDIO_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( RTAUDIO_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( RTAUDIO_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( &currentWritePointer, &safeWritePointer );
+ if ( FAILED( result ) ) {
+ errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current write position!";
+ errorText_ = errorStream_.str();
+ MUTEX_UNLOCK( &stream_.mutex );
+ error( RTAUDIO_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( &currentWritePointer, &safeWritePointer );
+ if ( FAILED( result ) ) {
+ errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current write position!";
+ errorText_ = errorStream_.str();
+ MUTEX_UNLOCK( &stream_.mutex );
+ error( RTAUDIO_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( RTAUDIO_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( RTAUDIO_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( &currentReadPointer, &safeReadPointer );
+ if ( FAILED( result ) ) {
+ errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current read position!";
+ errorText_ = errorStream_.str();
+ MUTEX_UNLOCK( &stream_.mutex );
+ error( RTAUDIO_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 aggressive.
+ 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( &currentReadPointer, &safeReadPointer );
+ if ( FAILED( result ) ) {
+ errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current read position!";
+ errorText_ = errorStream_.str();
+ MUTEX_UNLOCK( &stream_.mutex );
+ error( RTAUDIO_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( RTAUDIO_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( RTAUDIO_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 lpctstr,
+ 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 ( validDevice ) {
+ // If good device, then save its name and guid.
+ DsDevice device;
+ device.name = convertCharPointerToStdString( description );
+ device.epID = convertCharPointerToStdString(lpctstr);
+ device.id = lpguid;
+ device.isInput = probeInfo.isInput;
+ 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()
+#if _cplusplus >= 201103L
+ :handles{nullptr, nullptr}, synchronized(false), runnable(false) { xrun[0] = false; xrun[1] = false; }
+#else
+ : synchronized(false), runnable(false) { handles[0] = NULL; handles[1] = NULL; xrun[0] = false; xrun[1] = false; }
+#endif
+};
+
+static void *alsaCallbackHandler( void * ptr );
+
+RtApiAlsa :: RtApiAlsa()
+{
+ // Nothing to do here.
+}
+
+RtApiAlsa :: ~RtApiAlsa()
+{
+ if ( stream_.state != STREAM_CLOSED ) closeStream();
+}
+
+void RtApiAlsa :: probeDevices( void )
+{
+ // See list of required functionality in RtApi::probeDevices().
+
+ int result, device, card;
+ char name[128];
+ snd_ctl_t *handle = 0;
+ snd_ctl_card_info_t *ctlinfo;
+ snd_pcm_info_t *pcminfo;
+ snd_ctl_card_info_alloca(&ctlinfo);
+ snd_pcm_info_alloca(&pcminfo);
+ // First element isthe device hw ID, second is the device "pretty name"
+ std::vector<std::pair<std::string, std::string>> deviceID_prettyName;
+ snd_pcm_stream_t stream;
+ std::string defaultDeviceName;
+
+ // Add the default interface if available.
+ result = snd_ctl_open( &handle, "default", 0 );
+ if (result == 0) {
+ deviceID_prettyName.push_back({"default", "Default ALSA Device"});
+ defaultDeviceName = deviceID_prettyName[0].second;
+ snd_ctl_close( handle );
+ }
+
+ // Add the Pulse interface if available.
+ result = snd_ctl_open( &handle, "pulse", 0 );
+ if (result == 0) {
+ deviceID_prettyName.push_back({"pulse", "PulseAudio Sound Server"});
+ snd_ctl_close( handle );
+ }
+
+ // Count cards and devices and get ascii identifiers.
+ card = -1;
+ snd_card_next( &card );
+ while ( card >= 0 ) {
+ sprintf( name, "hw:%d", card );
+ result = snd_ctl_open( &handle, name, 0 );
+ if ( result < 0 ) {
+ handle = 0;
+ errorStream_ << "RtApiAlsa::probeDevices: control open, card = " << card << ", " << snd_strerror( result ) << ".";
+ errorText_ = errorStream_.str();
+ error( RTAUDIO_WARNING );
+ goto nextcard;
+ }
+ result = snd_ctl_card_info( handle, ctlinfo );
+ if ( result < 0 ) {
+ errorStream_ << "RtApiAlsa::probeDevices: control info, card = " << card << ", " << snd_strerror( result ) << ".";
+ errorText_ = errorStream_.str();
+ error( RTAUDIO_WARNING );
+ goto nextcard;
+ }
+ device = -1;
+ while( 1 ) {
+ result = snd_ctl_pcm_next_device( handle, &device );
+ if ( result < 0 ) {
+ errorStream_ << "RtApiAlsa::probeDevices: control next device, card = " << card << ", " << snd_strerror( result ) << ".";
+ errorText_ = errorStream_.str();
+ error( RTAUDIO_WARNING );
+ break;
+ }
+ if ( device < 0 )
+ break;
+
+ snd_pcm_info_set_device( pcminfo, device );
+ snd_pcm_info_set_subdevice( pcminfo, 0 );
+ stream = SND_PCM_STREAM_PLAYBACK;
+ snd_pcm_info_set_stream( pcminfo, stream );
+ result = snd_ctl_pcm_info( handle, pcminfo );
+ if ( result < 0 ) {
+ if ( result == -ENOENT ) { // try as input stream
+ stream = SND_PCM_STREAM_CAPTURE;
+ snd_pcm_info_set_stream( pcminfo, stream );
+ result = snd_ctl_pcm_info( handle, pcminfo );
+ if ( result < 0 ) {
+ errorStream_ << "RtApiAlsa::probeDevices: control pcm info, card = " << card << ", device = " << device << ", " << snd_strerror( result ) << ".";
+ errorText_ = errorStream_.str();
+ error( RTAUDIO_WARNING );
+ continue;
+ }
+ }
+ else continue;
+ }
+ sprintf( name, "hw:%s,%d", snd_ctl_card_info_get_id(ctlinfo), device );
+ std::string id(name);
+ sprintf( name, "%s (%s)", snd_ctl_card_info_get_name(ctlinfo), snd_pcm_info_get_id(pcminfo) );
+ std::string prettyName(name);
+ deviceID_prettyName.push_back( {id, prettyName} );
+ if ( card == 0 && device == 0 && defaultDeviceName.empty() )
+ defaultDeviceName = name;
+ }
+ nextcard:
+ if ( handle )
+ snd_ctl_close( handle );
+ snd_card_next( &card );
+ }
+
+ if ( deviceID_prettyName.size() == 0 ) {
+ deviceList_.clear();
+ deviceIdPairs_.clear();
+ return;
+ }
+
+ // Clean removed devices
+ for ( auto it = deviceIdPairs_.begin(); it != deviceIdPairs_.end(); ) {
+ bool found = false;
+ for ( auto& d: deviceID_prettyName ) {
+ if ( d.first == (*it).first ) {
+ found = true;
+ break;
+ }
+ }
+
+ if ( found )
+ ++it;
+ else
+ it = deviceIdPairs_.erase(it);
+ }
+
+ // Fill or update the deviceList_ and also save a corresponding list of Ids.
+ for ( auto& d : deviceID_prettyName ) {
+ bool found = false;
+ for ( auto& dID : deviceIdPairs_ ) {
+ if ( d.first == dID.first ) {
+ found = true;
+ break; // We already have this device.
+ }
+ }
+
+ if ( found )
+ continue;
+
+ // new device
+ RtAudio::DeviceInfo info;
+ info.name = d.second;
+ if ( probeDeviceInfo( info, d.first ) == false ) continue; // ignore if probe fails
+ info.ID = currentDeviceId_++; // arbitrary internal device ID
+ if ( info.name == defaultDeviceName ) {
+ if ( info.outputChannels > 0 ) info.isDefaultOutput = true;
+ if ( info.inputChannels > 0 ) info.isDefaultInput = true;
+ }
+ deviceList_.push_back( info );
+ deviceIdPairs_.push_back({d.first, info.ID});
+ // I don't see that ALSA provides property listeners to know if
+ // devices are removed or added.
+ }
+
+ // Remove any devices left in the list that are no longer available.
+ for ( std::vector<RtAudio::DeviceInfo>::iterator it=deviceList_.begin(); it!=deviceList_.end(); )
+ {
+ auto itID = deviceIdPairs_.begin();
+ while ( itID != deviceIdPairs_.end() ) {
+ if ( (*it).ID == (*itID).second ) {
+ break;
+ }
+ ++itID;
+ }
+
+ if ( itID == deviceIdPairs_.end() ) {
+ // not found so remove it from our list
+ it = deviceList_.erase( it );
+ }
+ else
+ ++it;
+ }
+}
+
+bool RtApiAlsa :: probeDeviceInfo( RtAudio::DeviceInfo& info, std::string name )
+{
+ int result, openMode = SND_PCM_ASYNC;
+ snd_pcm_stream_t stream;
+ snd_pcm_t *phandle;
+ snd_pcm_hw_params_t *params;
+ snd_pcm_hw_params_alloca( &params );
+
+ // First try for playback
+ stream = SND_PCM_STREAM_PLAYBACK;
+ result = snd_pcm_open( &phandle, name.c_str(), stream, openMode | SND_PCM_NONBLOCK );
+ if ( result < 0 ) {
+ if ( result == -16 ) return false; // device busy ... can't probe or use
+ if ( result != -2 ) { // device doesn't support playback
+ errorStream_ << "RtApiAlsa::probeDeviceInfo: snd_pcm_open (playback) error for device (" << name << "), " << snd_strerror( result ) << ".";
+ errorText_ = errorStream_.str();
+ error( RTAUDIO_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::probeDeviceInfo: snd_pcm_hw_params error for device (" << name << "), " << snd_strerror( result ) << ".";
+ errorText_ = errorStream_.str();
+ error( RTAUDIO_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::probeDeviceInfo: error getting device (" << name << ") output channels, " << snd_strerror( result ) << ".";
+ errorText_ = errorStream_.str();
+ error( RTAUDIO_WARNING );
+ goto captureProbe;
+ }
+ info.outputChannels = value;
+ snd_pcm_close( phandle );
+
+ captureProbe:
+ stream = SND_PCM_STREAM_CAPTURE;
+ result = snd_pcm_open( &phandle, name.c_str(), stream, openMode | SND_PCM_NONBLOCK);
+ if ( result < 0 && result ) {
+ if ( result != -2 && result != -16 ) { // device busy or doesn't support capture
+ errorStream_ << "RtApiAlsa::probeDeviceInfo: snd_pcm_open (capture) error for device (" << name << "), " << snd_strerror( result ) << ".";
+ errorText_ = errorStream_.str();
+ error( RTAUDIO_WARNING );
+ }
+ if ( info.outputChannels == 0 ) return false;
+ 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::probeDeviceInfo: snd_pcm_hw_params error for device (" << name << "), " << snd_strerror( result ) << ".";
+ errorText_ = errorStream_.str();
+ error( RTAUDIO_WARNING );
+ if ( info.outputChannels == 0 ) return false;
+ goto probeParameters;
+ }
+
+ result = snd_pcm_hw_params_get_channels_max( params, &value );
+ if ( result < 0 ) {
+ snd_pcm_close( phandle );
+ errorStream_ << "RtApiAlsa::probeDeviceInfo: error getting device (" << name << ") input channels, " << snd_strerror( result ) << ".";
+ errorText_ = errorStream_.str();
+ error( RTAUDIO_WARNING );
+ if ( info.outputChannels == 0 ) return false;
+ 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;
+
+ 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;
+
+ result = snd_pcm_open( &phandle, name.c_str(), stream, openMode | SND_PCM_NONBLOCK);
+ if ( result < 0 ) {
+ errorStream_ << "RtApiAlsa::probeDeviceInfo: snd_pcm_open error for device (" << name << "), " << snd_strerror( result ) << ".";
+ errorText_ = errorStream_.str();
+ error( RTAUDIO_WARNING );
+ return false;
+ }
+
+ // 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::probeDeviceInfo: snd_pcm_hw_params error for device (" << name << "), " << snd_strerror( result ) << ".";
+ errorText_ = errorStream_.str();
+ error( RTAUDIO_WARNING );
+ return false;
+ }
+
+ // 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::probeDeviceInfo: no supported sample rates found for device (" << name << ").";
+ errorText_ = errorStream_.str();
+ error( RTAUDIO_WARNING );
+ return false;
+ }
+
+ // 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::probeDeviceInfo: pcm device (" << name << ") data format not supported by RtAudio.";
+ errorText_ = errorStream_.str();
+ error( RTAUDIO_WARNING );
+ return false;
+ }
+
+ // Close the device and return
+ snd_pcm_close( phandle );
+ return true;
+}
+
+bool RtApiAlsa :: probeDeviceOpen( unsigned int deviceId, StreamMode mode, unsigned int channels,
+ unsigned int firstChannel, unsigned int sampleRate,
+ RtAudioFormat format, unsigned int *bufferSize,
+ RtAudio::StreamOptions *options )
+
+{
+#if defined(__RTAUDIO_DEBUG__)
+ struct SndOutputTdealloc {
+ SndOutputTdealloc() : _out(NULL) { snd_output_stdio_attach(&_out, stderr, 0); }
+ ~SndOutputTdealloc() { snd_output_close(_out); }
+ operator snd_output_t*() { return _out; }
+ snd_output_t *_out;
+ } out;
+#endif
+
+ std::string name;
+ for ( auto& id : deviceIdPairs_) {
+ if ( id.second == deviceId ) {
+ name = id.first;
+ break;
+ }
+ }
+
+ 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;
+ int result = snd_pcm_open( &phandle, name.c_str(), 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 (" << name << ") 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_.deviceId[mode] = deviceId;
+ 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( RTAUDIO_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 (e.g. NetBSD 1.6.x with GNU Pthread)
+ if ( options && options->flags & RTAUDIO_SCHEDULE_REALTIME ) {
+ stream_.callbackInfo.doRealtime = true;
+ 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;
+
+ // Set the policy BEFORE the priority. Otherwise it fails.
+ pthread_attr_setschedpolicy(&attr, SCHED_RR);
+ pthread_attr_setscope (&attr, PTHREAD_SCOPE_SYSTEM);
+ // This is definitely required. Otherwise it fails.
+ pthread_attr_setinheritsched(&attr, PTHREAD_EXPLICIT_SCHED);
+ pthread_attr_setschedparam(&attr, &param);
+ }
+ else
+ pthread_attr_setschedpolicy( &attr, SCHED_OTHER );
+#else
+ pthread_attr_setschedpolicy( &attr, SCHED_OTHER );
+#endif
+
+ stream_.callbackInfo.isRunning = true;
+ result = pthread_create( &stream_.callbackInfo.thread, &attr, alsaCallbackHandler, &stream_.callbackInfo );
+ pthread_attr_destroy( &attr );
+ if ( result ) {
+ // Failed. Try instead with default attributes.
+ result = pthread_create( &stream_.callbackInfo.thread, NULL, alsaCallbackHandler, &stream_.callbackInfo );
+ 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( RTAUDIO_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;
+ }
+
+ clearStreamInfo();
+}
+
+RtAudioErrorType RtApiAlsa :: startStream()
+{
+ // This method calls snd_pcm_prepare if the device isn't already in that state.
+
+ if ( stream_.state != STREAM_STOPPED ) {
+ if ( stream_.state == STREAM_RUNNING )
+ errorText_ = "RtApiAlsa::startStream(): the stream is already running!";
+ else if ( stream_.state == STREAM_STOPPING || stream_.state == STREAM_CLOSED )
+ errorText_ = "RtApiAlsa::startStream(): the stream is stopping or closed!";
+ return error( RTAUDIO_WARNING );
+ }
+
+ MUTEX_LOCK( &stream_.mutex );
+
+ /*
+ #if defined( HAVE_GETTIMEOFDAY )
+ gettimeofday( &stream_.lastTickTimestamp, NULL );
+ #endif
+ */
+
+ 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( RTAUDIO_SYSTEM_ERROR );
+ return RTAUDIO_NO_ERROR;
+}
+
+RtAudioErrorType RtApiAlsa :: stopStream()
+{
+ if ( stream_.state != STREAM_RUNNING && stream_.state != STREAM_STOPPING ) {
+ if ( stream_.state == STREAM_STOPPED )
+ errorText_ = "RtApiAlsa::stopStream(): the stream is already stopped!";
+ else if ( stream_.state == STREAM_CLOSED )
+ errorText_ = "RtApiAlsa::stopStream(): the stream is closed!";
+ return error( RTAUDIO_WARNING );
+ }
+
+ 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( RTAUDIO_SYSTEM_ERROR );
+ return RTAUDIO_NO_ERROR;
+}
+
+RtAudioErrorType RtApiAlsa :: abortStream()
+{
+ if ( stream_.state != STREAM_RUNNING ) {
+ if ( stream_.state == STREAM_STOPPED )
+ errorText_ = "RtApiAlsa::abortStream(): the stream is already stopped!";
+ else if ( stream_.state == STREAM_STOPPING || stream_.state == STREAM_CLOSED )
+ errorText_ = "RtApiAlsa::abortStream(): the stream is stopping or closed!";
+ return error( RTAUDIO_WARNING );
+ }
+
+ 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( RTAUDIO_SYSTEM_ERROR );
+ return RTAUDIO_NO_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( RTAUDIO_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 occurred.
+ 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( RTAUDIO_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 occurred.
+ 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( RTAUDIO_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 (e.g. NetBSD 1.6.x with GNU Pthread)
+ if ( info->doRealtime ) {
+ std::cerr << "RtAudio alsa: " <<
+ (sched_getscheduler(0) == SCHED_RR ? "" : "_NOT_ ") <<
+ "running realtime scheduling" << std::endl;
+ }
+#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.
+// Updated by Gary Scavone, 2021.
+
+#include <pulse/error.h>
+#include <pulse/simple.h>
+#include <cstdio>
+
+// A structure needed to pass variables for device probing.
+struct PaDeviceProbeInfo {
+ pa_mainloop_api *paMainLoopApi;
+ std::string defaultSinkName;
+ std::string defaultSourceName;
+ int defaultRate;
+ unsigned int *currentDeviceId;
+ std::vector< std::string > deviceNames;
+ std::vector< RtApiPulse::PaDeviceInfo > *paDeviceList;
+ std::vector< RtAudio::DeviceInfo > *rtDeviceList;
+};
+
+static const unsigned int SUPPORTED_SAMPLERATES[] = { 8000, 16000, 22050, 32000,
+ 44100, 48000, 96000, 192000, 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_SINT24, PA_SAMPLE_S24LE},
+ {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) { }
+};
+
+// The following 3 functions are called by the device probing
+// system. This first one gets overall system information.
+static void rt_pa_set_server_info( pa_context *context, const pa_server_info *info, void *userdata )
+{
+ (void)context;
+ pa_sample_spec ss;
+
+ PaDeviceProbeInfo *paProbeInfo = static_cast<PaDeviceProbeInfo *>( userdata );
+ if (!info) {
+ paProbeInfo->paMainLoopApi->quit( paProbeInfo->paMainLoopApi, 1 );
+ return;
+ }
+
+ ss = info->sample_spec;
+ paProbeInfo->defaultRate = ss.rate;
+ paProbeInfo->defaultSinkName = info->default_sink_name;
+ paProbeInfo->defaultSourceName = info->default_source_name;
+}
+
+// Used to get output device information.
+static void rt_pa_set_sink_info( pa_context * /*c*/, const pa_sink_info *i,
+ int eol, void *userdata )
+{
+ if ( eol ) return;
+
+ PaDeviceProbeInfo *paProbeInfo = static_cast<PaDeviceProbeInfo *>( userdata );
+ std::string name = pa_proplist_gets( i->proplist, "device.description" );
+ paProbeInfo->deviceNames.push_back( name );
+ for ( size_t n=0; n<paProbeInfo->rtDeviceList->size(); n++ )
+ if ( paProbeInfo->rtDeviceList->at(n).name == name ) return; // we've already probed this one
+
+ RtAudio::DeviceInfo info;
+ info.name = name;
+ info.outputChannels = i->sample_spec.channels;
+ info.preferredSampleRate = i->sample_spec.rate;
+ info.isDefaultOutput = ( paProbeInfo->defaultSinkName == i->name );
+ for ( const unsigned int *sr = SUPPORTED_SAMPLERATES; *sr; ++sr )
+ info.sampleRates.push_back( *sr );
+ for ( const rtaudio_pa_format_mapping_t *fm = supported_sampleformats; fm->rtaudio_format; ++fm )
+ info.nativeFormats |= fm->rtaudio_format;
+ info.ID = *(paProbeInfo->currentDeviceId);
+ *(paProbeInfo->currentDeviceId) = info.ID + 1;
+ paProbeInfo->rtDeviceList->push_back( info );
+
+ RtApiPulse::PaDeviceInfo painfo;
+ painfo.sinkName = i->name;
+ paProbeInfo->paDeviceList->push_back( painfo );
+}
+
+// Used to get input device information.
+static void rt_pa_set_source_info_and_quit( pa_context * /*c*/, const pa_source_info *i,
+ int eol, void *userdata )
+{
+ PaDeviceProbeInfo *paProbeInfo = static_cast<PaDeviceProbeInfo *>( userdata );
+ if ( eol ) {
+ paProbeInfo->paMainLoopApi->quit( paProbeInfo->paMainLoopApi, 0 );
+ return;
+ }
+
+ std::string name = pa_proplist_gets( i->proplist, "device.description" );
+ paProbeInfo->deviceNames.push_back( name );
+ for ( size_t n=0; n<paProbeInfo->rtDeviceList->size(); n++ ) {
+ if ( paProbeInfo->rtDeviceList->at(n).name == name ) {
+ // Check if we've already probed this as an output.
+ if ( !paProbeInfo->paDeviceList->at(n).sinkName.empty() ) {
+ // This must be a duplex device. Update the device info.
+ paProbeInfo->paDeviceList->at(n).sourceName = i->name;
+ paProbeInfo->rtDeviceList->at(n).inputChannels = i->sample_spec.channels;
+ paProbeInfo->rtDeviceList->at(n).isDefaultInput = ( paProbeInfo->defaultSourceName == i->name );
+ paProbeInfo->rtDeviceList->at(n).duplexChannels =
+ (paProbeInfo->rtDeviceList->at(n).inputChannels < paProbeInfo->rtDeviceList->at(n).outputChannels)
+ ? paProbeInfo->rtDeviceList->at(n).inputChannels : paProbeInfo->rtDeviceList->at(n).outputChannels;
+ }
+ return; // we already have this
+ }
+ }
+
+ RtAudio::DeviceInfo info;
+ info.name = name;
+ info.inputChannels = i->sample_spec.channels;
+ info.preferredSampleRate = i->sample_spec.rate;
+ info.isDefaultInput = ( paProbeInfo->defaultSourceName == i->name );
+ for ( const unsigned int *sr = SUPPORTED_SAMPLERATES; *sr; ++sr )
+ info.sampleRates.push_back( *sr );
+ for ( const rtaudio_pa_format_mapping_t *fm = supported_sampleformats; fm->rtaudio_format; ++fm )
+ info.nativeFormats |= fm->rtaudio_format;
+ info.ID = *(paProbeInfo->currentDeviceId);
+ *(paProbeInfo->currentDeviceId) = info.ID + 1;
+ paProbeInfo->rtDeviceList->push_back( info );
+
+ RtApiPulse::PaDeviceInfo painfo;
+ painfo.sourceName = i->name;
+ paProbeInfo->paDeviceList->push_back( painfo );
+}
+
+// This is the initial function that is called when the callback is
+// set. This one then calls the functions above.
+static void rt_pa_context_state_callback( pa_context *context, void *userdata )
+{
+ PaDeviceProbeInfo *paProbeInfo = static_cast<PaDeviceProbeInfo *>( userdata );
+ auto state = pa_context_get_state(context);
+ switch (state) {
+ case PA_CONTEXT_CONNECTING:
+ case PA_CONTEXT_AUTHORIZING:
+ case PA_CONTEXT_SETTING_NAME:
+ break;
+
+ case PA_CONTEXT_READY:
+ pa_context_get_server_info( context, rt_pa_set_server_info, userdata ); // server info
+ pa_context_get_sink_info_list( context, rt_pa_set_sink_info, userdata ); // output info ... needs to be before input
+ pa_context_get_source_info_list( context, rt_pa_set_source_info_and_quit, userdata ); // input info
+ break;
+
+ case PA_CONTEXT_TERMINATED:
+ paProbeInfo->paMainLoopApi->quit( paProbeInfo->paMainLoopApi, 0 );
+ break;
+
+ case PA_CONTEXT_FAILED:
+ default:
+ paProbeInfo->paMainLoopApi->quit( paProbeInfo->paMainLoopApi, 1 );
+ }
+}
+
+RtApiPulse::~RtApiPulse()
+{
+ if ( stream_.state != STREAM_CLOSED )
+ closeStream();
+}
+
+void RtApiPulse :: probeDevices( void )
+{
+ // See list of required functionality in RtApi::probeDevices().
+
+ pa_mainloop *ml = NULL;
+ pa_context *context = NULL;
+ char *server = NULL;
+ int ret = 1;
+ PaDeviceProbeInfo paProbeInfo;
+ if (!(ml = pa_mainloop_new())) {
+ errorStream_ << "RtApiPulse::probeDevices: pa_mainloop_new() failed.";
+ errorText_ = errorStream_.str();
+ error( RTAUDIO_WARNING );
+ goto quit;
+ }
+
+ paProbeInfo.paMainLoopApi = pa_mainloop_get_api( ml );
+ paProbeInfo.currentDeviceId = &currentDeviceId_;
+ paProbeInfo.paDeviceList = &paDeviceList_;
+ paProbeInfo.rtDeviceList = &deviceList_;
+
+ if (!(context = pa_context_new_with_proplist( paProbeInfo.paMainLoopApi, NULL, NULL ))) {
+ errorStream_ << "RtApiPulse::probeDevices: pa_context_new() failed.";
+ errorText_ = errorStream_.str();
+ error( RTAUDIO_WARNING );
+ goto quit;
+ }
+
+ pa_context_set_state_callback( context, rt_pa_context_state_callback, &paProbeInfo );
+
+ if (pa_context_connect( context, server, PA_CONTEXT_NOFLAGS, NULL ) < 0) {
+ errorStream_ << "RtApiPulse::probeDevices: pa_context_connect() failed: "
+ << pa_strerror(pa_context_errno(context));
+ errorText_ = errorStream_.str();
+ error( RTAUDIO_WARNING );
+ goto quit;
+ }
+
+ if (pa_mainloop_run( ml, &ret ) < 0) {
+ errorStream_ << "RtApiPulse::probeDevices: pa_mainloop_run() failed.";
+ errorText_ = errorStream_.str();
+ error( RTAUDIO_WARNING );
+ goto quit;
+ }
+
+ if (ret != 0) {
+ errorStream_ << "RtApiPulse::probeDevices: could not get server info.";
+ errorText_ = errorStream_.str();
+ error( RTAUDIO_WARNING );
+ goto quit;
+ }
+
+ // Check for devices that have been unplugged.
+ unsigned int m;
+ for ( std::vector<RtAudio::DeviceInfo>::iterator it=deviceList_.begin(); it!=deviceList_.end(); ) {
+ for ( m=0; m<paProbeInfo.deviceNames.size(); m++ ) {
+ if ( (*it).name == paProbeInfo.deviceNames[m] ) {
+ ++it;
+ break;
+ }
+ }
+ if ( m == paProbeInfo.deviceNames.size() ) { // not found so remove it from our list
+ it = deviceList_.erase( it );
+ paDeviceList_.erase( paDeviceList_.begin() + distance(deviceList_.begin(), it ) );
+ }
+ }
+
+quit:
+ if (context)
+ pa_context_unref(context);
+
+ if (ml)
+ pa_mainloop_free(ml);
+
+ pa_xfree(server);
+}
+
+static void *pulseaudio_callback( void * user )
+{
+ CallbackInfo *cbi = static_cast<CallbackInfo *>( user );
+ RtApiPulse *context = static_cast<RtApiPulse *>( cbi->object );
+ volatile bool *isRunning = &cbi->isRunning;
+
+#ifdef SCHED_RR // Undefined with some OSes (e.g. NetBSD 1.6.x with GNU Pthread)
+ if (cbi->doRealtime) {
+ std::cerr << "RtAudio pulse: " <<
+ (sched_getscheduler(0) == SCHED_RR ? "" : "_NOT_ ") <<
+ "running realtime scheduling" << std::endl;
+ }
+#endif
+
+ while ( *isRunning ) {
+ pthread_testcancel();
+ context->callbackEvent();
+ }
+
+ pthread_exit( NULL );
+}
+
+bool RtApiPulse::probeDeviceOpen( unsigned int deviceId, 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;
+
+ int deviceIdx = -1;
+ for ( unsigned int m=0; m<deviceList_.size(); m++ ) {
+ if ( deviceList_[m].ID == deviceId ) {
+ deviceIdx = m;
+ break;
+ }
+ }
+
+ if ( deviceIdx < 0 ) return false;
+
+ if ( firstChannel != 0 ) {
+ errorText_ = "PulseAudio does not support channel offset mapping.";
+ return false;
+ }
+
+ // These may be NULL for default devices but we already have the names.
+ const char *dev_input = NULL;
+ const char *dev_output = NULL;
+ if ( !paDeviceList_[deviceIdx].sourceName.empty() )
+ dev_input = paDeviceList_[deviceIdx].sourceName.c_str();
+ if ( !paDeviceList_[deviceIdx].sinkName.empty() )
+ dev_output = paDeviceList_[deviceIdx].sinkName.c_str();
+
+ if ( mode==INPUT && deviceList_[deviceIdx].inputChannels < channels ) {
+ errorText_ = "PulseAudio device does not support requested input channel count.";
+ return false;
+ }
+ if ( mode==OUTPUT && deviceList_[deviceIdx].outputChannels < channels ) {
+ errorText_ = "PulseAudio device does not support requested output channel count.";
+ return false;
+ }
+
+ ss.channels = channels;
+
+ 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 ) {
+ stream_.sampleRate = sampleRate;
+ ss.rate = sampleRate;
+ }
+
+ 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 = options ? options->numberOfBuffers : 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;
+ if ( stream_.userInterleaved != stream_.deviceInterleaved[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_.deviceId[mode] = deviceIdx;
+
+ // 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 ) {
+ pa_buffer_attr buffer_attr;
+ case INPUT:
+ buffer_attr.fragsize = bufferBytes;
+ buffer_attr.maxlength = -1;
+
+ pah->s_rec = pa_simple_new( NULL, streamName.c_str(), PA_STREAM_RECORD,
+ dev_input, "Record", &ss, NULL, &buffer_attr, &error );
+ if ( !pah->s_rec ) {
+ errorText_ = "RtApiPulse::probeDeviceOpen: error connecting input to PulseAudio server.";
+ goto error;
+ }
+ break;
+ case OUTPUT: {
+ pa_buffer_attr * attr_ptr;
+
+ if ( options && options->numberOfBuffers > 0 ) {
+ // pa_buffer_attr::fragsize is recording-only.
+ // Hopefully PortAudio won't access uninitialized fields.
+ buffer_attr.maxlength = bufferBytes * options->numberOfBuffers;
+ buffer_attr.minreq = -1;
+ buffer_attr.prebuf = -1;
+ buffer_attr.tlength = -1;
+ attr_ptr = &buffer_attr;
+ } else {
+ attr_ptr = nullptr;
+ }
+
+ pah->s_play = pa_simple_new( NULL, streamName.c_str(), PA_STREAM_PLAYBACK,
+ dev_output, "Playback", &ss, NULL, attr_ptr, &error );
+ if ( !pah->s_play ) {
+ errorText_ = "RtApiPulse::probeDeviceOpen: error connecting output to PulseAudio server.";
+ goto error;
+ }
+ break;
+ }
+ case DUPLEX:
+ /* Note: We could add DUPLEX by synchronizing multiple streams,
+ but it would mean moving from Simple API to Asynchronous API:
+ https://freedesktop.org/software/pulseaudio/doxygen/streams.html#sync_streams */
+ errorText_ = "RtApiPulse::probeDeviceOpen: duplex not supported for PulseAudio.";
+ goto error;
+ 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_.state = STREAM_STOPPED;
+ // 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 (e.g. NetBSD 1.6.x with GNU Pthread)
+ if ( options && options->flags & RTAUDIO_SCHEDULE_REALTIME ) {
+ stream_.callbackInfo.doRealtime = true;
+ 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;
+
+ // Set the policy BEFORE the priority. Otherwise it fails.
+ pthread_attr_setschedpolicy(&attr, SCHED_RR);
+ pthread_attr_setscope (&attr, PTHREAD_SCOPE_SYSTEM);
+ // This is definitely required. Otherwise it fails.
+ pthread_attr_setinheritsched(&attr, PTHREAD_EXPLICIT_SCHED);
+ pthread_attr_setschedparam(&attr, &param);
+ }
+ else
+ pthread_attr_setschedpolicy( &attr, SCHED_OTHER );
+#else
+ pthread_attr_setschedpolicy( &attr, SCHED_OTHER );
+#endif
+
+ stream_.callbackInfo.isRunning = true;
+ int result = pthread_create( &pah->thread, &attr, pulseaudio_callback, (void *)&stream_.callbackInfo);
+ pthread_attr_destroy(&attr);
+ if(result != 0) {
+ // Failed. Try instead with default attributes.
+ result = pthread_create( &pah->thread, NULL, pulseaudio_callback, (void *)&stream_.callbackInfo);
+ if(result != 0) {
+ stream_.callbackInfo.isRunning = false;
+ errorText_ = "RtApiPulse::probeDeviceOpen: error creating thread.";
+ goto error;
+ }
+ }
+ }
+
+ return SUCCESS;
+
+ 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;
+ }
+
+ stream_.state = STREAM_CLOSED;
+ return FAILURE;
+}
+
+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;
+ }
+
+ clearStreamInfo();
+}
+
+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( RTAUDIO_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( RTAUDIO_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( RTAUDIO_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();
+}
+
+RtAudioErrorType RtApiPulse::startStream( void )
+{
+ if ( stream_.state != STREAM_STOPPED ) {
+ if ( stream_.state == STREAM_RUNNING )
+ errorText_ = "RtApiPulse::startStream(): the stream is already running!";
+ else if ( stream_.state == STREAM_STOPPING || stream_.state == STREAM_CLOSED )
+ errorText_ = "RtApiPulse::startStream(): the stream is stopping or closed!";
+ return error( RTAUDIO_WARNING );
+ }
+
+ PulseAudioHandle *pah = static_cast<PulseAudioHandle *>( stream_.apiHandle );
+
+ MUTEX_LOCK( &stream_.mutex );
+
+ /*
+ #if defined( HAVE_GETTIMEOFDAY )
+ gettimeofday( &stream_.lastTickTimestamp, NULL );
+ #endif
+ */
+
+ stream_.state = STREAM_RUNNING;
+
+ pah->runnable = true;
+ pthread_cond_signal( &pah->runnable_cv );
+ MUTEX_UNLOCK( &stream_.mutex );
+ return RTAUDIO_NO_ERROR;
+}
+
+RtAudioErrorType RtApiPulse::stopStream( void )
+{
+ if ( stream_.state != STREAM_RUNNING && stream_.state != STREAM_STOPPING ) {
+ if ( stream_.state == STREAM_STOPPED )
+ errorText_ = "RtApiPulse::stopStream(): the stream is already stopped!";
+ else if ( stream_.state == STREAM_CLOSED )
+ errorText_ = "RtApiPulse::stopStream(): the stream is closed!";
+ return error( RTAUDIO_WARNING );
+ }
+
+ PulseAudioHandle *pah = static_cast<PulseAudioHandle *>( stream_.apiHandle );
+
+ stream_.state = STREAM_STOPPED;
+ MUTEX_LOCK( &stream_.mutex );
+
+ if ( pah ) {
+ pah->runnable = false;
+ if ( 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 );
+ return error( RTAUDIO_SYSTEM_ERROR );
+ }
+ }
+ }
+
+ stream_.state = STREAM_STOPPED;
+ MUTEX_UNLOCK( &stream_.mutex );
+ return RTAUDIO_NO_ERROR;
+}
+
+RtAudioErrorType RtApiPulse::abortStream( void )
+{
+ if ( stream_.state != STREAM_RUNNING ) {
+ if ( stream_.state == STREAM_STOPPED )
+ errorText_ = "RtApiPulse::abortStream(): the stream is already stopped!";
+ else if ( stream_.state == STREAM_STOPPING || stream_.state == STREAM_CLOSED )
+ errorText_ = "RtApiPulse::abortStream(): the stream is stopping or closed!";
+ return error( RTAUDIO_WARNING );
+ }
+
+ PulseAudioHandle *pah = static_cast<PulseAudioHandle*>( stream_.apiHandle );
+
+ stream_.state = STREAM_STOPPED;
+ MUTEX_LOCK( &stream_.mutex );
+
+ if ( pah ) {
+ pah->runnable = false;
+ if ( 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 );
+ return error( RTAUDIO_SYSTEM_ERROR );
+ }
+ }
+ }
+
+ stream_.state = STREAM_STOPPED;
+ MUTEX_UNLOCK( &stream_.mutex );
+ return RTAUDIO_NO_ERROR;
+}
+
+//******************** End of __LINUX_PULSE__ *********************//
+#endif
+
+#if defined(__LINUX_OSS__)
+
+#include <unistd.h>
+#include <sys/ioctl.h>
+#include <unistd.h>
+#include <fcntl.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();
+}
+
+void RtApiOss :: probeDevices( void )
+{
+ // See list of required functionality in RtApi::probeDevices().
+
+ int mixerfd = open( "/dev/mixer", O_RDWR, 0 );
+ if ( mixerfd == -1 ) {
+ errorText_ = "RtApiOss::probeDevices: error opening '/dev/mixer'.";
+ error( RTAUDIO_SYSTEM_ERROR );
+ return;
+ }
+
+ oss_sysinfo sysinfo;
+ if ( ioctl( mixerfd, SNDCTL_SYSINFO, &sysinfo ) == -1 ) {
+ close( mixerfd );
+ errorText_ = "RtApiOss::probeDevices: error getting sysinfo, OSS version >= 4.0 is required.";
+ error( RTAUDIO_SYSTEM_ERROR );
+ return;
+ }
+
+ unsigned int nDevices = sysinfo.numaudios;
+ if ( nDevices == 0 ) {
+ close( mixerfd );
+ deviceList_.clear();
+ return;
+ }
+
+ oss_audioinfo ainfo;
+ unsigned int m, n;
+ std::vector<std::string> deviceNames;
+ for ( n=0; n<nDevices; n++ ) {
+ ainfo.dev = n;
+ if ( ioctl( mixerfd, SNDCTL_AUDIOINFO, &ainfo ) == -1 ) continue;
+ deviceNames.push_back( ainfo.name );
+ for ( m=0; m<deviceList_.size(); m++ ) {
+ if ( deviceList_[m].name == deviceNames.back() )
+ break; // We already have this device.
+ }
+ if ( m == deviceList_.size() ) { // new device
+ RtAudio::DeviceInfo info;
+ if ( probeDeviceInfo( info, ainfo ) == false ) continue; // ignore if probe fails
+ info.ID = currentDeviceId_++; // arbitrary internal device ID
+ deviceList_.push_back( info );
+ }
+ }
+ close( mixerfd );
+
+ // Remove any devices left in the list that are no longer available.
+ for ( std::vector<RtAudio::DeviceInfo>::iterator it=deviceList_.begin(); it!=deviceList_.end(); ) {
+ for ( m=0; m<deviceNames.size(); m++ ) {
+ if ( (*it).name == deviceNames[m] ) {
+ ++it;
+ break;
+ }
+ }
+ if ( m == deviceNames.size() ) // not found so remove it from our list
+ it = deviceList_.erase( it );
+ }
+
+ // I don't think the OSS API supports default devices. Our parent
+ // class versions of the getDefault functions will return the first
+ // one found.
+}
+
+bool RtApiOss :: probeDeviceInfo( RtAudio::DeviceInfo &info, oss_audioinfo &ainfo )
+{
+ // 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::probeDeviceInfo: device (" << ainfo.name << ") data format not supported by RtAudio.";
+ errorText_ = errorStream_.str();
+ error( RTAUDIO_WARNING );
+ return false;
+ }
+
+ // 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::probeDeviceInfo: no supported sample rates found for device (" << ainfo.name << ").";
+ errorText_ = errorStream_.str();
+ error( RTAUDIO_WARNING );
+ return false;
+ }
+
+ return true;
+}
+
+
+bool RtApiOss :: probeDeviceOpen( unsigned int deviceId, 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 int 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;
+ }
+
+ std::string deviceName;
+ unsigned int m, device;
+ for ( m=0; m<deviceList_.size(); m++ ) {
+ if ( deviceList_[m].ID == deviceId ) {
+ deviceName = deviceList_[m].name;
+ break;
+ }
+ }
+
+ if ( deviceName.empty() ) {
+ errorText_ = "RtApiOss::probeDeviceOpen: device ID is invalid!";
+ return FAILURE;
+ }
+
+ oss_audioinfo ainfo;
+ for ( device=0; device<nDevices; device++ ) {
+ ainfo.dev = device;
+ result = ioctl( mixerfd, SNDCTL_AUDIOINFO, &ainfo );
+ if ( result == -1 ) continue;
+ if ( deviceName == std::string( ainfo.name ) ) break;
+ }
+
+ close( mixerfd );
+ if ( device == nDevices ) {
+ errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") not found.";
+ 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_.deviceId[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_.deviceId[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_.deviceId[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_.deviceId[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 (e.g. NetBSD 1.6.x with GNU Pthread)
+ if ( options && options->flags & RTAUDIO_SCHEDULE_REALTIME ) {
+ stream_.callbackInfo.doRealtime = true;
+ 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;
+
+ // Set the policy BEFORE the priority. Otherwise it fails.
+ pthread_attr_setschedpolicy(&attr, SCHED_RR);
+ pthread_attr_setscope (&attr, PTHREAD_SCOPE_SYSTEM);
+ // This is definitely required. Otherwise it fails.
+ pthread_attr_setinheritsched(&attr, PTHREAD_EXPLICIT_SCHED);
+ pthread_attr_setschedparam(&attr, &param);
+ }
+ 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 ) {
+ // Failed. Try instead with default attributes.
+ result = pthread_create( &stream_.callbackInfo.thread, NULL, ossCallbackHandler, &stream_.callbackInfo );
+ 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;
+ }
+
+ stream_.state = STREAM_CLOSED;
+ return FAILURE;
+}
+
+void RtApiOss :: closeStream()
+{
+ if ( stream_.state == STREAM_CLOSED ) {
+ errorText_ = "RtApiOss::closeStream(): no open stream to close!";
+ error( RTAUDIO_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;
+ }
+
+ clearStreamInfo();
+ //stream_.mode = UNINITIALIZED;
+ //stream_.state = STREAM_CLOSED;
+}
+
+RtAudioErrorType RtApiOss :: startStream()
+{
+ if ( stream_.state != STREAM_STOPPED ) {
+ if ( stream_.state == STREAM_RUNNING )
+ errorText_ = "RtApiOss::startStream(): the stream is already running!";
+ else if ( stream_.state == STREAM_STOPPING || stream_.state == STREAM_CLOSED )
+ errorText_ = "RtApiOss::startStream(): the stream is stopping or closed!";
+ return error( RTAUDIO_WARNING );
+ }
+
+ MUTEX_LOCK( &stream_.mutex );
+
+ /*
+ #if defined( HAVE_GETTIMEOFDAY )
+ gettimeofday( &stream_.lastTickTimestamp, NULL );
+ #endif
+ */
+
+ 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 );
+ return RTAUDIO_NO_ERROR;
+}
+
+RtAudioErrorType RtApiOss :: stopStream()
+{
+ if ( stream_.state != STREAM_RUNNING && stream_.state != STREAM_STOPPING ) {
+ if ( stream_.state == STREAM_STOPPED )
+ errorText_ = "RtApiOss::stopStream(): the stream is already stopped!";
+ else if ( stream_.state == STREAM_CLOSED )
+ errorText_ = "RtApiOss::stopStream(): the stream is closed!";
+ return error( RTAUDIO_WARNING );
+ }
+
+ MUTEX_LOCK( &stream_.mutex );
+
+ // The state might change while waiting on a mutex.
+ if ( stream_.state == STREAM_STOPPED ) {
+ MUTEX_UNLOCK( &stream_.mutex );
+ return RTAUDIO_NO_ERROR;
+ }
+
+ 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( RTAUDIO_WARNING );
+ }
+ }
+
+ result = ioctl( handle->id[0], SNDCTL_DSP_HALT, 0 );
+ if ( result == -1 ) {
+ errorStream_ << "RtApiOss::stopStream: system error stopping callback procedure on device (" << stream_.deviceId[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_.deviceId[0] << ").";
+ errorText_ = errorStream_.str();
+ goto unlock;
+ }
+ }
+
+ unlock:
+ stream_.state = STREAM_STOPPED;
+ MUTEX_UNLOCK( &stream_.mutex );
+
+ if ( result != -1 ) return RTAUDIO_NO_ERROR;
+ return error( RTAUDIO_SYSTEM_ERROR );
+}
+
+RtAudioErrorType RtApiOss :: abortStream()
+{
+ if ( stream_.state != STREAM_RUNNING ) {
+ if ( stream_.state == STREAM_STOPPED )
+ errorText_ = "RtApiOss::abortStream(): the stream is already stopped!";
+ else if ( stream_.state == STREAM_STOPPING || stream_.state == STREAM_CLOSED )
+ errorText_ = "RtApiOss::abortStream(): the stream is stopping or closed!";
+ return error( RTAUDIO_WARNING );
+ }
+
+ MUTEX_LOCK( &stream_.mutex );
+
+ // The state might change while waiting on a mutex.
+ if ( stream_.state == STREAM_STOPPED ) {
+ MUTEX_UNLOCK( &stream_.mutex );
+ return RTAUDIO_NO_ERROR;
+ }
+
+ 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_.deviceId[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_.deviceId[0] << ").";
+ errorText_ = errorStream_.str();
+ goto unlock;
+ }
+ }
+
+ unlock:
+ stream_.state = STREAM_STOPPED;
+ MUTEX_UNLOCK( &stream_.mutex );
+
+ if ( result != -1 ) return RTAUDIO_SYSTEM_ERROR;
+ return error( RTAUDIO_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( RTAUDIO_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( RTAUDIO_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( RTAUDIO_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;
+
+#ifdef SCHED_RR // Undefined with some OSes (e.g. NetBSD 1.6.x with GNU Pthread)
+ if (info->doRealtime) {
+ std::cerr << "RtAudio oss: " <<
+ (sched_getscheduler(0) == SCHED_RR ? "" : "_NOT_ ") <<
+ "running realtime scheduling" << std::endl;
+ }
+#endif
+
+ 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.
+RtAudioErrorType RtApi :: error( RtAudioErrorType type )
+{
+ errorStream_.str(""); // clear the ostringstream to avoid repeated messages
+
+ // Don't output warnings if showWarnings_ is false
+ if ( type == RTAUDIO_WARNING && showWarnings_ == false ) return type;
+
+ if ( errorCallback_ ) {
+ //const std::string errorMessage = errorText_;
+ //errorCallback_( type, errorMessage );
+ errorCallback_( type, errorText_ );
+ }
+ else
+ std::cerr << '\n' << errorText_ << "\n\n";
+ return 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.deviceDisconnected = false;
+ for ( int i=0; i<2; i++ ) {
+ stream_.deviceId[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( RTAUDIO_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 duplex device output buffer if there are more device outputs than user outputs
+ if ( outBuffer == stream_.deviceBuffer && stream_.mode == DUPLEX && info.outJump > info.inJump )
+ memset( outBuffer, 0, stream_.bufferSize * info.outJump * formatBytes( info.outFormat ) );
+
+ int j;
+ if (info.outFormat == RTAUDIO_FLOAT64) {
+ Float64 *out = (Float64 *)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]] = (Float64) in[info.inOffset[j]] / 128.0;
+ }
+ 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]] = (Float64) in[info.inOffset[j]] / 32768.0;
+ }
+ 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]] = (Float64) in[info.inOffset[j]].asInt() / 8388608.0;
+ }
+ 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]] = (Float64) in[info.inOffset[j]] / 2147483648.0;
+ }
+ 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 *out = (Float32 *)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]] = (Float32) in[info.inOffset[j]] / 128.f;
+ }
+ 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]] = (Float32) in[info.inOffset[j]] / 32768.f;
+ }
+ 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]] = (Float32) in[info.inOffset[j]].asInt() / 8388608.f;
+ }
+ 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]] = (Float32) in[info.inOffset[j]] / 2147483648.f;
+ }
+ 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++) {
+ // Use llround() which returns `long long` which is guaranteed to be at least 64 bits.
+ out[info.outOffset[j]] = (Int32) std::max(std::min(std::llround(in[info.inOffset[j]] * 2147483648.f), 2147483647LL), -2147483648LL);
+ }
+ 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) std::max(std::min(std::llround(in[info.inOffset[j]] * 2147483648.0), 2147483647LL), -2147483648LL);
+ }
+ 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) std::max(std::min(std::llround(in[info.inOffset[j]] * 8388608.f), 8388607LL), -8388608LL);
+ }
+ 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) std::max(std::min(std::llround(in[info.inOffset[j]] * 8388608.0), 8388607LL), -8388608LL);
+ }
+ 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) std::max(std::min(std::llround(in[info.inOffset[j]] * 32768.f), 32767LL), -32768LL);
+ }
+ 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) std::max(std::min(std::llround(in[info.inOffset[j]] * 32768.0), 32767LL), -32768LL);
+ }
+ 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) std::max(std::min(std::llround(in[info.inOffset[j]] * 128.f), 127LL), -128LL);
+ }
+ 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) std::max(std::min(std::llround(in[info.inOffset[j]] * 128.0), 127LL), -128LL);
+ }
+ 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;
+ }
+ }
+}
+
+ // Indentation settings for Vim and Emacs
+ //
+ // Local Variables:
+ // c-basic-offset: 2
+ // indent-tabs-mode: nil
+ // End:
+ //
+ // vim: et sts=2 sw=2
+
diff --git a/src/modules/rtaudio/RtAudio.h b/src/modules/rtaudio/RtAudio.h
index 7de8d03c..e767ddb2 100644
--- a/src/modules/rtaudio/RtAudio.h
+++ b/src/modules/rtaudio/RtAudio.h
@@ -7,10 +7,11 @@
and OSS), Macintosh OS X (CoreAudio and Jack), and Windows
(DirectSound, ASIO and WASAPI) operating systems.
+ RtAudio GitHub site: https://github.com/thestk/rtaudio
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) 2001-2023 Gary P. Scavone
Permission is hereby granted, free of charge, to any person
obtaining a copy of this software and associated documentation files
@@ -45,12 +46,43 @@
#ifndef __RTAUDIO_H
#define __RTAUDIO_H
-#define RTAUDIO_VERSION "4.1.2"
+#define RTAUDIO_VERSION_MAJOR 6
+#define RTAUDIO_VERSION_MINOR 0
+#define RTAUDIO_VERSION_PATCH 1
+#define RTAUDIO_VERSION_BETA 0
+
+#define RTAUDIO_TOSTRING2(n) #n
+#define RTAUDIO_TOSTRING(n) RTAUDIO_TOSTRING2(n)
+
+#if RTAUDIO_VERSION_BETA > 0
+ #define RTAUDIO_VERSION RTAUDIO_TOSTRING(RTAUDIO_VERSION_MAJOR) \
+ "." RTAUDIO_TOSTRING(RTAUDIO_VERSION_MINOR) \
+ "." RTAUDIO_TOSTRING(RTAUDIO_VERSION_PATCH) \
+ "beta" RTAUDIO_TOSTRING(RTAUDIO_VERSION_BETA)
+#else
+ #define RTAUDIO_VERSION RTAUDIO_TOSTRING(RTAUDIO_VERSION_MAJOR) \
+ "." RTAUDIO_TOSTRING(RTAUDIO_VERSION_MINOR) \
+ "." RTAUDIO_TOSTRING(RTAUDIO_VERSION_PATCH)
+#endif
+
+#if defined _WIN32 || defined __CYGWIN__
+ #if defined(RTAUDIO_EXPORT)
+ #define RTAUDIO_DLL_PUBLIC __declspec(dllexport)
+ #else
+ #define RTAUDIO_DLL_PUBLIC
+ #endif
+#else
+ #if __GNUC__ >= 4
+ #define RTAUDIO_DLL_PUBLIC __attribute__( (visibility( "default" )) )
+ #else
+ #define RTAUDIO_DLL_PUBLIC
+ #endif
+#endif
#include <string>
#include <vector>
-#include <exception>
#include <iostream>
+#include <functional>
/*! \typedef typedef unsigned long RtAudioFormat;
\brief RtAudio data format type.
@@ -60,6 +92,8 @@
internal routines will automatically take care of any necessary
byte-swapping between the host format and the soundcard. Thus,
endian-ness is not a concern in the following format definitions.
+ Note that there are no range checks for floating-point values that
+ extend beyond plus/minus 1.0.
- \e RTAUDIO_SINT8: 8-bit signed integer.
- \e RTAUDIO_SINT16: 16-bit signed integer.
@@ -86,6 +120,7 @@ static const RtAudioFormat RTAUDIO_FLOAT64 = 0x20; // Normalized between plus/mi
- \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
@@ -101,7 +136,7 @@ static const RtAudioFormat RTAUDIO_FLOAT64 = 0x20; // Normalized between plus/mi
Certain audio APIs offer a number of parameters that influence the
I/O latency of a stream. By default, RtAudio will attempt to set
these parameters internally for robust (glitch-free) performance
- (though some APIs, like Windows Direct Sound, make this difficult).
+ (though some APIs, like Windows DirectSound, make this difficult).
By passing the RTAUDIO_MINIMIZE_LATENCY flag to the openStream()
function, internal stream settings will be influenced in an attempt
to minimize stream latency, though possibly at the expense of stream
@@ -117,6 +152,9 @@ static const RtAudioFormat RTAUDIO_FLOAT64 = 0x20; // Normalized between plus/mi
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 +162,7 @@ static const RtAudioStreamFlags RTAUDIO_MINIMIZE_LATENCY = 0x2; // Attempt to s
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.
@@ -174,6 +213,7 @@ static const RtAudioStreamStatus RTAUDIO_OUTPUT_UNDERFLOW = 0x2; // The output
\param userData A pointer to optional data provided by the client
when opening the stream (default = NULL).
+ \return
To continue normal stream operation, the RtAudioCallback function
should return a value of zero. To stop the stream and drain the
output buffer, the function should return a value of one. To abort
@@ -185,55 +225,19 @@ typedef int (*RtAudioCallback)( void *outputBuffer, void *inputBuffer,
RtAudioStreamStatus status,
void *userData );
-/************************************************************************/
-/*! \class RtAudioError
- \brief Exception handling class for RtAudio.
-
- The RtAudioError class is quite simple but it does allow errors to be
- "caught" by RtAudioError::Type. See the RtAudio documentation to know
- which methods can throw an RtAudioError.
-*/
-/************************************************************************/
-
-class RtAudioError : public std::exception
-{
- public:
- //! Defined RtAudioError types.
- enum Type {
- WARNING, /*!< A non-critical error. */
- DEBUG_WARNING, /*!< A non-critical error which might be useful for debugging. */
- UNSPECIFIED, /*!< The default, unspecified error type. */
- NO_DEVICES_FOUND, /*!< No devices found on system. */
- INVALID_DEVICE, /*!< An invalid device ID was specified. */
- MEMORY_ERROR, /*!< An error occurred during memory allocation. */
- INVALID_PARAMETER, /*!< An invalid parameter was specified to a function. */
- INVALID_USE, /*!< The function was called incorrectly. */
- DRIVER_ERROR, /*!< A system driver error occurred. */
- SYSTEM_ERROR, /*!< A system error occurred. */
- THREAD_ERROR /*!< A thread error occurred. */
- };
-
- //! The constructor.
- RtAudioError( const std::string& message, Type type = RtAudioError::UNSPECIFIED ) throw() : message_(message), type_(type) {}
-
- //! The destructor.
- virtual ~RtAudioError( void ) throw() {}
-
- //! Prints thrown error message to stderr.
- virtual void printMessage( void ) const throw() { std::cerr << '\n' << message_ << "\n\n"; }
-
- //! Returns the thrown error message type.
- virtual const Type& getType(void) const throw() { 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(); }
-
- protected:
- std::string message_;
- Type type_;
+enum RtAudioErrorType {
+ RTAUDIO_NO_ERROR = 0, /*!< No error. */
+ RTAUDIO_WARNING, /*!< A non-critical error. */
+ RTAUDIO_UNKNOWN_ERROR, /*!< An unspecified error type. */
+ RTAUDIO_NO_DEVICES_FOUND, /*!< No devices found on system. */
+ RTAUDIO_INVALID_DEVICE, /*!< An invalid device ID was specified. */
+ RTAUDIO_DEVICE_DISCONNECT, /*!< A device in use was disconnected. */
+ RTAUDIO_MEMORY_ERROR, /*!< An error occurred during memory allocation. */
+ RTAUDIO_INVALID_PARAMETER, /*!< An invalid parameter was specified to a function. */
+ RTAUDIO_INVALID_USE, /*!< The function was called incorrectly. */
+ RTAUDIO_DRIVER_ERROR, /*!< A system driver error occurred. */
+ RTAUDIO_SYSTEM_ERROR, /*!< A system error occurred. */
+ RTAUDIO_THREAD_ERROR /*!< A thread error occurred. */
};
//! RtAudio error callback function prototype.
@@ -241,7 +245,9 @@ class RtAudioError : public std::exception
\param type Type of error.
\param errorText Error description.
*/
-typedef void (*RtAudioErrorCallback)( RtAudioError::Type type, const std::string &errorText );
+typedef std::function<void(RtAudioErrorType type,
+ const std::string &errorText )>
+ RtAudioErrorCallback;
// **************************************************************** //
//
@@ -258,52 +264,46 @@ typedef void (*RtAudioErrorCallback)( RtAudioError::Type type, const std::string
class RtApi;
-class RtAudio
+class RTAUDIO_DLL_PUBLIC RtAudio
{
public:
//! Audio API specifier arguments.
enum Api {
UNSPECIFIED, /*!< Search for a working compiled API. */
+ MACOSX_CORE, /*!< Macintosh OS-X Core Audio API. */
LINUX_ALSA, /*!< The Advanced Linux Sound Architecture API. */
+ UNIX_JACK, /*!< The Jack Low-Latency Audio Server API. */
LINUX_PULSE, /*!< The Linux PulseAudio API. */
LINUX_OSS, /*!< The Linux Open Sound System API. */
- UNIX_JACK, /*!< The Jack Low-Latency Audio Server API. */
- MACOSX_CORE, /*!< Macintosh OS-X Core Audio API. */
- WINDOWS_WASAPI, /*!< The Microsoft WASAPI API. */
WINDOWS_ASIO, /*!< The Steinberg Audio Stream I/O API. */
- WINDOWS_DS, /*!< The Microsoft Direct Sound API. */
- RTAUDIO_DUMMY /*!< A compilable but non-functional API. */
+ WINDOWS_WASAPI, /*!< The Microsoft WASAPI API. */
+ WINDOWS_DS, /*!< The Microsoft DirectSound API. */
+ RTAUDIO_DUMMY, /*!< A compilable but non-functional API. */
+ NUM_APIS /*!< Number of values in this enum. */
};
//! The public device information structure for returning queried values.
struct DeviceInfo {
- bool probed; /*!< true if the device capabilities were successfully probed. */
- std::string name; /*!< Character string device identifier. */
- unsigned int outputChannels; /*!< Maximum output channels supported by device. */
- unsigned int inputChannels; /*!< Maximum input channels supported by device. */
- unsigned int duplexChannels; /*!< Maximum simultaneous input/output channels supported by device. */
- bool isDefaultOutput; /*!< true if this is the default output device. */
- bool isDefaultInput; /*!< true if this is the default input device. */
+ unsigned int ID{}; /*!< Device ID used to specify a device to RtAudio. */
+ std::string name; /*!< Character string device name. */
+ unsigned int outputChannels{}; /*!< Maximum output channels supported by device. */
+ unsigned int inputChannels{}; /*!< Maximum input channels supported by device. */
+ unsigned int duplexChannels{}; /*!< Maximum simultaneous input/output channels supported by device. */
+ bool isDefaultOutput{false}; /*!< true if this is the default output device. */
+ bool isDefaultInput{false}; /*!< true if this is the default input device. */
std::vector<unsigned int> sampleRates; /*!< Supported sample rates (queried from list of standard rates). */
- unsigned int preferredSampleRate; /*!< Preferred sample rate, eg. for WASAPI the system sample rate. */
- RtAudioFormat nativeFormats; /*!< Bit mask of supported data formats. */
-
- // Default constructor.
- DeviceInfo()
- :probed(false), outputChannels(0), inputChannels(0), duplexChannels(0),
- isDefaultOutput(false), isDefaultInput(false), preferredSampleRate(0), nativeFormats(0) {}
+ unsigned int currentSampleRate{}; /*!< Current sample rate, system sample rate as currently configured. */
+ unsigned int preferredSampleRate{}; /*!< Preferred sample rate, e.g. for WASAPI the system sample rate. */
+ RtAudioFormat nativeFormats{}; /*!< Bit mask of supported data formats. */
};
//! The structure for specifying input or output stream parameters.
struct StreamParameters {
- unsigned int deviceId; /*!< Device index (0 to getDeviceCount() - 1). */
- unsigned int nChannels; /*!< Number of channels. */
- unsigned int firstChannel; /*!< First channel index on device (default = 0). */
-
- // Default constructor.
- StreamParameters()
- : deviceId(0), nChannels(0), firstChannel(0) {}
+ //std::string deviceName{}; /*!< Device name from device list. */
+ unsigned int deviceId{}; /*!< Device id as provided by getDeviceIds(). */
+ unsigned int nChannels{}; /*!< Number of channels. */
+ unsigned int firstChannel{}; /*!< First channel index on device (default = 0). */
};
//! The structure for specifying stream options.
@@ -331,7 +331,7 @@ class RtAudio
Certain audio APIs offer a number of parameters that influence the
I/O latency of a stream. By default, RtAudio will attempt to set
these parameters internally for robust (glitch-free) performance
- (though some APIs, like Windows Direct Sound, make this difficult).
+ (though some APIs, like Windows DirectSound, make this difficult).
By passing the RTAUDIO_MINIMIZE_LATENCY flag to the openStream()
function, internal stream settings will be influenced in an attempt
to minimize stream latency, though possibly at the expense of stream
@@ -359,23 +359,21 @@ class RtAudio
function by the value actually used by the system.
The \c streamName parameter can be used to set the client name
- when using the Jack API. By default, the client name is set to
- RtApiJack. However, if you wish to create multiple instances of
- RtAudio with Jack, each instance must have a unique client name.
+ when using the Jack API or the application name when using the
+ Pulse API. By default, the Jack client name is set to RtApiJack.
+ However, if you wish to create multiple instances of RtAudio with
+ Jack, each instance must have a unique client name. The default
+ Pulse application name is set to "RtAudio."
*/
struct StreamOptions {
- RtAudioStreamFlags flags; /*!< A bit-mask of stream flags (RTAUDIO_NONINTERLEAVED, RTAUDIO_MINIMIZE_LATENCY, RTAUDIO_HOG_DEVICE, RTAUDIO_ALSA_USE_DEFAULT). */
- unsigned int numberOfBuffers; /*!< Number of stream buffers. */
+ RtAudioStreamFlags flags{}; /*!< A bit-mask of stream flags (RTAUDIO_NONINTERLEAVED, RTAUDIO_MINIMIZE_LATENCY, RTAUDIO_HOG_DEVICE, RTAUDIO_ALSA_USE_DEFAULT). */
+ unsigned int numberOfBuffers{}; /*!< Number of stream buffers. */
std::string streamName; /*!< A stream name (currently used only in Jack). */
- int priority; /*!< Scheduling priority of callback thread (only used with flag RTAUDIO_SCHEDULE_REALTIME). */
-
- // Default constructor.
- StreamOptions()
- : flags(0), numberOfBuffers(0), priority(0) {}
+ int priority{}; /*!< Scheduling priority of callback thread (only used with flag RTAUDIO_SCHEDULE_REALTIME). */
};
//! 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.
/*!
@@ -383,90 +381,152 @@ class RtAudio
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.
+ //! Return the name of a specified compiled audio API.
+ /*!
+ This obtains a short lower-case name used for identification purposes.
+ This value is guaranteed to remain identical across library versions.
+ If the API is unknown, this function will return the empty string.
+ */
+ static std::string getApiName( RtAudio::Api api );
+
+ //! Return the display name of a specified compiled audio API.
+ /*!
+ This obtains a long name used for display purposes.
+ If the API is unknown, this function will return the empty string.
+ */
+ static std::string getApiDisplayName( RtAudio::Api api );
+
+ //! Return the compiled audio API having the given name.
/*!
- The constructor performs minor initialization tasks. An exception
- can be thrown if no API support is compiled.
+ A case insensitive comparison will check the specified name
+ against the list of compiled APIs, and return the one that
+ matches. On failure, the function returns UNSPECIFIED.
+ */
+ static RtAudio::Api getCompiledApiByName( const std::string &name );
- If no API argument is specified and multiple API support has been
- compiled, the default order of use is JACK, ALSA, OSS (Linux
- systems) and ASIO, DS (Windows systems).
+ //! Return the compiled audio API having the given display name.
+ /*!
+ A case sensitive comparison will check the specified display name
+ against the list of compiled APIs, and return the one that
+ matches. On failure, the function returns UNSPECIFIED.
*/
- RtAudio( RtAudio::Api api=UNSPECIFIED );
+ static RtAudio::Api getCompiledApiByDisplayName( const std::string &name );
+
+ //! The class constructor.
+ /*!
+ The constructor attempts to create an RtApi instance.
+
+ If an API argument is specified but that API has not been
+ compiled, a warning is issued and an instance of an available API
+ is created. If no compiled API is found, the routine will abort
+ (though this should be impossible because RtDummy is the default
+ if no API-specific preprocessor definition is provided to the
+ compiler). If no API argument is specified and multiple API
+ support has been compiled, the default order of use is JACK, ALSA,
+ OSS (Linux systems) and ASIO, DS (Windows systems).
+
+ An optional errorCallback function can be specified to
+ subsequently receive warning and error messages.
+ */
+ RtAudio( RtAudio::Api api=UNSPECIFIED, RtAudioErrorCallback&& errorCallback=0 );
//! The destructor.
/*!
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.
+ This function performs a system query of available devices each
+ time it is called, thus supporting devices (dis)connected \e after
+ instantiation. If a system error occurs during processing, a
+ warning will be issued.
+ */
+ unsigned int getDeviceCount( void );
+
+ //! A public function that returns a vector of audio device IDs.
+ /*!
+ The ID values returned by this function are used internally by
+ RtAudio to identify a given device. The values themselves are
+ arbitrary and do not correspond to device IDs used by the
+ underlying API (nor are they index values). This function performs
+ a system query of available devices each time it is called, thus
+ supporting devices (dis)connected \e after instantiation. If no
+ devices are available, the vector size will be zero. If a system
+ error occurs during processing, a warning will be issued.
*/
- unsigned int getDeviceCount( void ) throw();
+ std::vector<unsigned int> getDeviceIds( void );
- //! Return an RtAudio::DeviceInfo structure for a specified device number.
+ //! A public function that returns a vector of audio device names.
/*!
+ This function performs a system query of available devices each
+ time it is called, thus supporting devices (dis)connected \e after
+ instantiation. If no devices are available, the vector size will
+ be zero. If a system error occurs during processing, a warning
+ will be issued.
+ */
+ std::vector<std::string> getDeviceNames( void );
- Any device integer between 0 and getDeviceCount() - 1 is valid.
- If an invalid argument is provided, an RtAudioError (type = INVALID_USE)
- will be thrown. If a device is busy or otherwise unavailable, the
- structure member "probed" will have a value of "false" and all
- other members are undefined. If the specified device is the
- current default input or output device, the corresponding
- "isDefault" member will have a value of "true".
+ //! Return an RtAudio::DeviceInfo structure for a specified device ID.
+ /*!
+ Any device ID returned by getDeviceIds() is valid, unless it has
+ been removed between the call to getDevceIds() and this
+ function. If an invalid argument is provided, an
+ RTAUDIO_INVALID_USE will be passed to the user-provided
+ errorCallback function (or otherwise printed to stderr) and all
+ members of the returned RtAudio::DeviceInfo structure will be
+ initialized to default, invalid values (ID = 0, empty name, ...).
+ If the specified device is the current default input or output
+ device, the corresponding "isDefault" member will have a value of
+ "true".
*/
- RtAudio::DeviceInfo getDeviceInfo( unsigned int device );
+ RtAudio::DeviceInfo getDeviceInfo( unsigned int deviceId );
- //! A function that returns the index of the default output device.
+ //! A function that returns the ID of the default output device.
/*!
- If the underlying audio API does not provide a "default
- device", or if no devices are available, the return value will be
- 0. Note that this is a valid device identifier and it is the
- client's responsibility to verify that a device is available
- before attempting to open a stream.
+ If the underlying audio API does not provide a "default device",
+ the first probed output device ID will be returned. If no devices
+ are available, the return value will be 0 (which is an invalid
+ device identifier).
*/
- unsigned int getDefaultOutputDevice( void ) throw();
+ unsigned int getDefaultOutputDevice( void );
- //! A function that returns the index of the default input device.
+ //! A function that returns the ID of the default input device.
/*!
- If the underlying audio API does not provide a "default
- device", or if no devices are available, the return value will be
- 0. Note that this is a valid device identifier and it is the
- client's responsibility to verify that a device is available
- before attempting to open a stream.
+ If the underlying audio API does not provide a "default device",
+ the first probed input device ID will be returned. If no devices
+ are available, the return value will be 0 (which is an invalid
+ device identifier).
*/
- unsigned int getDefaultInputDevice( void ) throw();
+ unsigned int getDefaultInputDevice( void );
//! A public function for opening a stream with the specified parameters.
/*!
- An RtAudioError (type = SYSTEM_ERROR) is thrown if a stream cannot be
+ An RTAUDIO_SYSTEM_ERROR is returned if a stream cannot be
opened with the specified parameters or an error occurs during
- processing. An RtAudioError (type = INVALID_USE) is thrown if any
- invalid device ID or channel number parameters are specified.
+ processing. An RTAUDIO_INVALID_USE is returned if a stream
+ is already open or any invalid stream parameters are specified.
\param outputParameters Specifies output stream parameters to use
when opening a stream, including a device ID, number of channels,
and starting channel number. For input-only streams, this
- argument should be NULL. The device ID is an index value between
- 0 and getDeviceCount() - 1.
+ argument should be NULL. The device ID is a value returned by
+ getDeviceIds().
\param inputParameters Specifies input stream parameters to use
when opening a stream, including a device ID, number of channels,
and starting channel number. For output-only streams, this
- argument should be NULL. The device ID is an index value between
- 0 and getDeviceCount() - 1.
+ argument should be NULL. The device ID is a value returned by
+ getDeviceIds().
\param format An RtAudioFormat specifying the desired sample data format.
\param sampleRate The desired sample rate (sample frames per second).
- \param *bufferFrames A pointer to a value indicating the desired
+ \param bufferFrames A pointer to a value indicating the desired
internal buffer size in sample frames. The actual value
used by the device is returned via the same pointer. A
value of zero can be specified, in which case the lowest
@@ -484,65 +544,70 @@ class RtAudio
chosen. If the RTAUDIO_MINIMIZE_LATENCY flag bit is set, the
lowest allowable value is used. The actual value used is
returned via the structure argument. The parameter is API dependent.
- \param errorCallback A client-defined function that will be invoked
- when an error has occurred.
*/
- void openStream( RtAudio::StreamParameters *outputParameters,
- RtAudio::StreamParameters *inputParameters,
- RtAudioFormat format, unsigned int sampleRate,
- unsigned int *bufferFrames, RtAudioCallback callback,
- void *userData = NULL, RtAudio::StreamOptions *options = NULL, RtAudioErrorCallback errorCallback = NULL );
+ RtAudioErrorType openStream( RtAudio::StreamParameters *outputParameters,
+ RtAudio::StreamParameters *inputParameters,
+ RtAudioFormat format, unsigned int sampleRate,
+ unsigned int *bufferFrames, RtAudioCallback callback,
+ void *userData = NULL, RtAudio::StreamOptions *options = NULL );
//! A function that closes a stream and frees any associated stream memory.
/*!
- If a stream is not open, this function issues a warning and
- returns (no exception is thrown).
+ If a stream is not open, an RTAUDIO_WARNING will be passed to the
+ user-provided errorCallback function (or otherwise printed to
+ stderr).
*/
- void closeStream( void ) throw();
+ void closeStream( void );
//! A function that starts a stream.
/*!
- An RtAudioError (type = SYSTEM_ERROR) is thrown if an error occurs
- during processing. An RtAudioError (type = INVALID_USE) is thrown if a
- stream is not open. A warning is issued if the stream is already
- running.
+ An RTAUDIO_SYSTEM_ERROR is returned if an error occurs during
+ processing. An RTAUDIO_WARNING is returned if a stream is not open
+ or is already running.
*/
- void startStream( void );
+ RtAudioErrorType startStream( void );
//! Stop a stream, allowing any samples remaining in the output queue to be played.
/*!
- An RtAudioError (type = SYSTEM_ERROR) is thrown if an error occurs
- during processing. An RtAudioError (type = INVALID_USE) is thrown if a
- stream is not open. A warning is issued if the stream is already
- stopped.
+ An RTAUDIO_SYSTEM_ERROR is returned if an error occurs during
+ processing. An RTAUDIO_WARNING is returned if a stream is not
+ open or is already stopped.
*/
- void stopStream( void );
+ RtAudioErrorType stopStream( void );
//! Stop a stream, discarding any samples remaining in the input/output queue.
/*!
- An RtAudioError (type = SYSTEM_ERROR) is thrown if an error occurs
- during processing. An RtAudioError (type = INVALID_USE) is thrown if a
- stream is not open. A warning is issued if the stream is already
- stopped.
+ An RTAUDIO_SYSTEM_ERROR is returned if an error occurs during
+ processing. An RTAUDIO_WARNING is returned if a stream is not
+ open or is already stopped.
+ */
+ RtAudioErrorType abortStream( void );
+
+ //! Retrieve the error message corresponding to the last error or warning condition.
+ /*!
+ This function can be used to get a detailed error message when a
+ non-zero RtAudioErrorType is returned by a function. This is the
+ same message sent to the user-provided errorCallback function.
*/
- void abortStream( void );
+ const std::string getErrorText( 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.
+ //! Returns the number of seconds of processed data since the stream was started.
/*!
- If a stream is not open, an RtAudioError (type = INVALID_USE) will be thrown.
+ The stream time is calculated from the number of sample frames
+ processed by the underlying audio system, which will increment by
+ units of the audio buffer size. It is not an absolute running
+ time. If a stream is not open, the returned value may not be
+ valid.
*/
double getStreamTime( void );
//! Set the stream time to a time in seconds greater than or equal to 0.0.
- /*!
- If a stream is not open, an RtAudioError (type = INVALID_USE) will be thrown.
- */
void setStreamTime( double time );
//! Returns the internal stream latency in sample frames.
@@ -550,22 +615,30 @@ class RtAudio
The stream latency refers to delay in audio input and/or output
caused by internal buffering by the audio system and/or hardware.
For duplex streams, the returned value will represent the sum of
- the input and output latencies. If a stream is not open, an
- RtAudioError (type = INVALID_USE) will be thrown. If the API does not
- report latency, the return value will be zero.
+ the input and output latencies. If a stream is not open, the
+ returned value will be invalid. If the API does not report
+ latency, the return value will be zero.
*/
long getStreamLatency( void );
- //! Returns actual sample rate in use by the stream.
- /*!
- On some systems, the sample rate used may be slightly different
- than that specified in the stream parameters. If a stream is not
- open, an RtAudioError (type = INVALID_USE) will be thrown.
- */
+ //! Returns actual sample rate in use by the (open) stream.
+ /*!
+ On some systems, the sample rate used may be slightly different
+ than that specified in the stream parameters. If a stream is not
+ open, a value of zero is returned.
+ */
unsigned int getStreamSampleRate( void );
- //! Specify whether warning messages should be printed to stderr.
- void showWarnings( bool value = true ) throw();
+ //! Set a client-defined function that will be invoked when an error or warning occurs.
+ void setErrorCallback( RtAudioErrorCallback errorCallback );
+
+ //! Specify whether warning messages should be output or not.
+ /*!
+ The default behaviour is for warning messages to be output,
+ either to a client-defined error callback function (if specified)
+ or to stderr.
+ */
+ void showWarnings( bool value = true );
protected:
@@ -574,29 +647,34 @@ class RtAudio
};
// Operating system dependent thread functionality.
-#if defined(__WINDOWS_DS__) || defined(__WINDOWS_ASIO__) || defined(__WINDOWS_WASAPI__)
+#if defined(_MSC_VER)
#ifndef NOMINMAX
#define NOMINMAX
#endif
#include <windows.h>
#include <process.h>
+ #include <stdint.h>
typedef uintptr_t ThreadHandle;
typedef CRITICAL_SECTION StreamMutex;
-#elif defined(__LINUX_ALSA__) || defined(__LINUX_PULSE__) || defined(__UNIX_JACK__) || defined(__LINUX_OSS__) || defined(__MACOSX_CORE__)
+#else
+
// Using pthread library for various flavors of unix.
#include <pthread.h>
typedef pthread_t ThreadHandle;
typedef pthread_mutex_t StreamMutex;
-#else // Setup for "dummy" behavior
+#endif
+
+// Setup for "dummy" behavior if no apis specified.
+#if !(defined(__WINDOWS_DS__) || defined(__WINDOWS_ASIO__) || defined(__WINDOWS_WASAPI__) \
+ || defined(__LINUX_ALSA__) || defined(__LINUX_PULSE__) || defined(__UNIX_JACK__) \
+ || defined(__LINUX_OSS__) || defined(__MACOSX_CORE__))
#define __RTAUDIO_DUMMY__
- typedef int ThreadHandle;
- typedef int StreamMutex;
#endif
@@ -604,19 +682,15 @@ class RtAudio
// between the private RtAudio stream structure and global callback
// handling functions.
struct CallbackInfo {
- void *object; // Used as a "this" pointer.
- ThreadHandle thread;
- void *callback;
- void *userData;
- void *errorCallback;
- void *apiInfo; // void pointer for API specific callback information
- bool isRunning;
- bool doRealtime;
- int priority;
-
- // Default constructor.
- CallbackInfo()
- :object(0), callback(0), userData(0), errorCallback(0), apiInfo(0), isRunning(false), doRealtime(false) {}
+ void *object{}; // Used as a "this" pointer.
+ ThreadHandle thread{};
+ void *callback{};
+ void *userData{};
+ void *apiInfo{}; // void pointer for API specific callback information
+ bool isRunning{false};
+ bool doRealtime{false};
+ int priority{};
+ bool deviceDisconnected{false};
};
// **************************************************************** //
@@ -643,13 +717,12 @@ class S24 {
S24() {}
S24& operator = ( const int& i ) {
- c3[0] = (i & 0x000000ff);
- c3[1] = (i & 0x0000ff00) >> 8;
- c3[2] = (i & 0x00ff0000) >> 16;
+ c3[0] = (unsigned char)(i & 0x000000ff);
+ c3[1] = (unsigned char)((i & 0x0000ff00) >> 8);
+ c3[2] = (unsigned char)((i & 0x00ff0000) >> 16);
return *this;
}
- S24( const S24& v ) { *this = v; }
S24( const double& d ) { *this = (int) d; }
S24( const float& f ) { *this = (int) f; }
S24( const signed short& s ) { *this = (int) s; }
@@ -669,33 +742,36 @@ class S24 {
#include <sstream>
-class RtApi
+class RTAUDIO_DLL_PUBLIC RtApi
{
public:
RtApi();
virtual ~RtApi();
virtual RtAudio::Api getCurrentApi( void ) = 0;
- virtual unsigned int getDeviceCount( void ) = 0;
- virtual RtAudio::DeviceInfo getDeviceInfo( unsigned int device ) = 0;
+ unsigned int getDeviceCount( void );
+ std::vector<unsigned int> getDeviceIds( void );
+ std::vector<std::string> getDeviceNames( void );
+ RtAudio::DeviceInfo getDeviceInfo( unsigned int deviceId );
virtual unsigned int getDefaultInputDevice( void );
virtual unsigned int getDefaultOutputDevice( void );
- void openStream( RtAudio::StreamParameters *outputParameters,
- RtAudio::StreamParameters *inputParameters,
- RtAudioFormat format, unsigned int sampleRate,
- unsigned int *bufferFrames, RtAudioCallback callback,
- void *userData, RtAudio::StreamOptions *options,
- RtAudioErrorCallback errorCallback );
+ RtAudioErrorType openStream( RtAudio::StreamParameters *outputParameters,
+ RtAudio::StreamParameters *inputParameters,
+ RtAudioFormat format, unsigned int sampleRate,
+ unsigned int *bufferFrames, RtAudioCallback callback,
+ void *userData, RtAudio::StreamOptions *options );
virtual void closeStream( void );
- virtual void startStream( void ) = 0;
- virtual void stopStream( void ) = 0;
- virtual void abortStream( void ) = 0;
+ virtual RtAudioErrorType startStream( void ) = 0;
+ virtual RtAudioErrorType stopStream( void ) = 0;
+ virtual RtAudioErrorType abortStream( void ) = 0;
+ const std::string getErrorText( void ) const { return errorText_; }
long getStreamLatency( void );
unsigned int getStreamSampleRate( void );
- virtual double getStreamTime( void );
+ virtual double getStreamTime( void ) const { return stream_.streamTime; }
virtual void setStreamTime( double time );
bool isStreamOpen( void ) const { return stream_.state != STREAM_CLOSED; }
bool isStreamRunning( void ) const { return stream_.state == STREAM_RUNNING; }
+ void setErrorCallback( RtAudioErrorCallback errorCallback ) { errorCallback_ = errorCallback; }
void showWarnings( bool value ) { showWarnings_ = value; }
@@ -731,7 +807,7 @@ protected:
// A protected structure for audio streams.
struct RtApiStream {
- unsigned int device[2]; // Playback and record, respectively.
+ unsigned int deviceId[2]; // Playback and record, respectively.
void *apiHandle; // void pointer for API specific stream handle information
StreamMode mode; // OUTPUT, INPUT, or DUPLEX.
StreamState state; // STOPPED, RUNNING, or CLOSED
@@ -760,7 +836,7 @@ protected:
#endif
RtApiStream()
- :apiHandle(0), deviceBuffer(0) { device[0] = 11111; device[1] = 11111; }
+ :apiHandle(0), deviceBuffer(0) {} // { device[0] = std::string(); device[1] = std::string(); }
};
typedef S24 Int24;
@@ -771,10 +847,22 @@ protected:
std::ostringstream errorStream_;
std::string errorText_;
+ RtAudioErrorCallback errorCallback_;
bool showWarnings_;
+ std::vector<RtAudio::DeviceInfo> deviceList_;
+ unsigned int currentDeviceId_;
RtApiStream stream_;
- bool firstErrorOccurred_;
+ /*!
+ Protected, api-specific method that attempts to probe all device
+ capabilities in a system. The function will not re-probe devices
+ that were previously found and probed. This function MUST be
+ implemented by all subclasses. If an error is encountered during
+ the probe, a "warning" message may be reported and the internal
+ list of devices may be incomplete.
+ */
+ virtual void probeDevices( void );
+
/*!
Protected, api-specific method that attempts to open a device
with the given parameters. This function MUST be implemented by
@@ -782,7 +870,7 @@ protected:
"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 deviceId, StreamMode mode, unsigned int channels,
unsigned int firstChannel, unsigned int sampleRate,
RtAudioFormat format, unsigned int *bufferSize,
RtAudio::StreamOptions *options );
@@ -793,14 +881,8 @@ protected:
//! Protected common method to clear an RtApiStream structure.
void clearStreamInfo();
- /*!
- Protected common method that throws an RtAudioError (type =
- INVALID_USE) if a stream is not open.
- */
- void verifyStream( void );
-
//! Protected common error method to allow global control over error handling.
- void error( RtAudioError::Type type );
+ RtAudioErrorType error( RtAudioErrorType type );
/*!
Protected method used to perform format, channel number, and/or interleaving
@@ -824,332 +906,26 @@ protected:
//
// **************************************************************** //
-inline RtAudio::Api RtAudio :: getCurrentApi( void ) throw() { return rtapi_->getCurrentApi(); }
-inline unsigned int RtAudio :: getDeviceCount( void ) throw() { 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 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 RtAudio::Api RtAudio :: getCurrentApi( void ) { return rtapi_->getCurrentApi(); }
+inline unsigned int RtAudio :: getDeviceCount( void ) { return rtapi_->getDeviceCount(); }
+inline RtAudio::DeviceInfo RtAudio :: getDeviceInfo( unsigned int deviceId ) { return rtapi_->getDeviceInfo( deviceId ); }
+inline std::vector<unsigned int> RtAudio :: getDeviceIds( void ) { return rtapi_->getDeviceIds(); }
+inline std::vector<std::string> RtAudio :: getDeviceNames( void ) { return rtapi_->getDeviceNames(); }
+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 RtAudioErrorType RtAudio :: startStream( void ) { return rtapi_->startStream(); }
+inline RtAudioErrorType RtAudio :: stopStream( void ) { return rtapi_->stopStream(); }
+inline RtAudioErrorType RtAudio :: abortStream( void ) { return rtapi_->abortStream(); }
+inline const std::string RtAudio :: getErrorText( void ) { return rtapi_->getErrorText(); }
+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 ); }
-
-// RtApi Subclass prototypes.
-
-#if defined(__MACOSX_CORE__)
-
-#include <CoreAudio/AudioHardware.h>
-
-class RtApiCore: public RtApi
-{
-public:
-
- RtApiCore();
- ~RtApiCore();
- RtAudio::Api getCurrentApi( void ) { return RtAudio::MACOSX_CORE; }
- unsigned int getDeviceCount( void );
- RtAudio::DeviceInfo getDeviceInfo( unsigned int device );
- unsigned int getDefaultOutputDevice( void );
- unsigned int getDefaultInputDevice( void );
- void closeStream( void );
- void startStream( void );
- void stopStream( void );
- void abortStream( void );
- long getStreamLatency( void );
-
- // This function is intended for internal use only. It must be
- // public because it is called by the internal callback handler,
- // which is not a member of RtAudio. External use of this function
- // will most likely produce highly undesirable results!
- bool callbackEvent( AudioDeviceID deviceId,
- const AudioBufferList *inBufferList,
- const AudioBufferList *outBufferList );
-
- private:
-
- bool probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,
- unsigned int firstChannel, unsigned int sampleRate,
- RtAudioFormat format, unsigned int *bufferSize,
- RtAudio::StreamOptions *options );
- static const char* getErrorCode( OSStatus code );
-};
-
-#endif
-
-#if defined(__UNIX_JACK__)
-
-class RtApiJack: public RtApi
-{
-public:
-
- RtApiJack();
- ~RtApiJack();
- RtAudio::Api getCurrentApi( void ) { return RtAudio::UNIX_JACK; }
- unsigned int getDeviceCount( void );
- RtAudio::DeviceInfo getDeviceInfo( unsigned int device );
- void closeStream( void );
- void startStream( void );
- void stopStream( void );
- void abortStream( void );
- long getStreamLatency( void );
-
- // This function is intended for internal use only. It must be
- // public because it is called by the internal callback handler,
- // which is not a member of RtAudio. External use of this function
- // will most likely produce highly undesirable results!
- bool callbackEvent( unsigned long nframes );
-
- private:
-
- bool probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,
- unsigned int firstChannel, unsigned int sampleRate,
- RtAudioFormat format, unsigned int *bufferSize,
- RtAudio::StreamOptions *options );
-};
-
-#endif
-
-#if defined(__WINDOWS_ASIO__)
-
-class RtApiAsio: public RtApi
-{
-public:
-
- RtApiAsio();
- ~RtApiAsio();
- RtAudio::Api getCurrentApi( void ) { return RtAudio::WINDOWS_ASIO; }
- unsigned int getDeviceCount( void );
- RtAudio::DeviceInfo getDeviceInfo( unsigned int device );
- void closeStream( void );
- void startStream( void );
- void stopStream( void );
- void abortStream( void );
- long getStreamLatency( void );
-
- // This function is intended for internal use only. It must be
- // public because it is called by the internal callback handler,
- // which is not a member of RtAudio. External use of this function
- // will most likely produce highly undesirable results!
- bool callbackEvent( long bufferIndex );
-
- private:
-
- std::vector<RtAudio::DeviceInfo> devices_;
- void saveDeviceInfo( void );
- bool coInitialized_;
- bool probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,
- unsigned int firstChannel, unsigned int sampleRate,
- RtAudioFormat format, unsigned int *bufferSize,
- RtAudio::StreamOptions *options );
-};
-
-#endif
-
-#if defined(__WINDOWS_DS__)
-
-class RtApiDs: public RtApi
-{
-public:
-
- RtApiDs();
- ~RtApiDs();
- RtAudio::Api getCurrentApi( void ) { return RtAudio::WINDOWS_DS; }
- unsigned int getDeviceCount( void );
- unsigned int getDefaultOutputDevice( void );
- unsigned int getDefaultInputDevice( void );
- RtAudio::DeviceInfo getDeviceInfo( unsigned int device );
- void closeStream( void );
- void startStream( void );
- void stopStream( void );
- void abortStream( void );
- long getStreamLatency( void );
-
- // This function is intended for internal use only. It must be
- // public because it is called by the internal callback handler,
- // which is not a member of RtAudio. External use of this function
- // will most likely produce highly undesirable results!
- void callbackEvent( void );
-
- private:
-
- bool coInitialized_;
- bool buffersRolling;
- long duplexPrerollBytes;
- std::vector<struct DsDevice> dsDevices;
- bool probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,
- unsigned int firstChannel, unsigned int sampleRate,
- RtAudioFormat format, unsigned int *bufferSize,
- RtAudio::StreamOptions *options );
-};
-
-#endif
-
-#if defined(__WINDOWS_WASAPI__)
-
-struct IMMDeviceEnumerator;
-
-class RtApiWasapi : public RtApi
-{
-public:
- RtApiWasapi();
- ~RtApiWasapi();
-
- RtAudio::Api getCurrentApi( void ) { return RtAudio::WINDOWS_WASAPI; }
- unsigned int getDeviceCount( void );
- RtAudio::DeviceInfo getDeviceInfo( unsigned int device );
- unsigned int getDefaultOutputDevice( void );
- unsigned int getDefaultInputDevice( void );
- void closeStream( void );
- void startStream( void );
- void stopStream( void );
- void abortStream( void );
-
-private:
- bool coInitialized_;
- IMMDeviceEnumerator* deviceEnumerator_;
-
- bool probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,
- unsigned int firstChannel, unsigned int sampleRate,
- RtAudioFormat format, unsigned int* bufferSize,
- RtAudio::StreamOptions* options );
-
- static DWORD WINAPI runWasapiThread( void* wasapiPtr );
- static DWORD WINAPI stopWasapiThread( void* wasapiPtr );
- static DWORD WINAPI abortWasapiThread( void* wasapiPtr );
- void wasapiThread();
-};
-
-#endif
-
-#if defined(__LINUX_ALSA__)
-
-class RtApiAlsa: public RtApi
-{
-public:
-
- RtApiAlsa();
- ~RtApiAlsa();
- RtAudio::Api getCurrentApi() { return RtAudio::LINUX_ALSA; }
- unsigned int getDeviceCount( void );
- RtAudio::DeviceInfo getDeviceInfo( unsigned int device );
- void closeStream( void );
- void startStream( void );
- void stopStream( void );
- void abortStream( void );
-
- // This function is intended for internal use only. It must be
- // public because it is called by the internal callback handler,
- // which is not a member of RtAudio. External use of this function
- // will most likely produce highly undesirable results!
- void callbackEvent( void );
-
- private:
-
- std::vector<RtAudio::DeviceInfo> devices_;
- void saveDeviceInfo( void );
- bool probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,
- unsigned int firstChannel, unsigned int sampleRate,
- RtAudioFormat format, unsigned int *bufferSize,
- RtAudio::StreamOptions *options );
-};
-
-#endif
-
-#if defined(__LINUX_PULSE__)
-
-class RtApiPulse: public RtApi
-{
-public:
- ~RtApiPulse();
- RtAudio::Api getCurrentApi() { return RtAudio::LINUX_PULSE; }
- unsigned int getDeviceCount( void );
- RtAudio::DeviceInfo getDeviceInfo( unsigned int device );
- void closeStream( void );
- void startStream( void );
- void stopStream( void );
- void abortStream( void );
-
- // This function is intended for internal use only. It must be
- // public because it is called by the internal callback handler,
- // which is not a member of RtAudio. External use of this function
- // will most likely produce highly undesirable results!
- void callbackEvent( void );
-
- private:
-
- std::vector<RtAudio::DeviceInfo> devices_;
- void saveDeviceInfo( void );
- bool probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,
- unsigned int firstChannel, unsigned int sampleRate,
- RtAudioFormat format, unsigned int *bufferSize,
- RtAudio::StreamOptions *options );
-};
-
-#endif
-
-#if defined(__LINUX_OSS__)
-
-class RtApiOss: public RtApi
-{
-public:
-
- RtApiOss();
- ~RtApiOss();
- RtAudio::Api getCurrentApi() { return RtAudio::LINUX_OSS; }
- unsigned int getDeviceCount( void );
- RtAudio::DeviceInfo getDeviceInfo( unsigned int device );
- void closeStream( void );
- void startStream( void );
- void stopStream( void );
- void abortStream( void );
-
- // This function is intended for internal use only. It must be
- // public because it is called by the internal callback handler,
- // which is not a member of RtAudio. External use of this function
- // will most likely produce highly undesirable results!
- void callbackEvent( void );
-
- private:
-
- bool probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,
- unsigned int firstChannel, unsigned int sampleRate,
- RtAudioFormat format, unsigned int *bufferSize,
- RtAudio::StreamOptions *options );
-};
-
-#endif
-
-#if defined(__RTAUDIO_DUMMY__)
-
-class RtApiDummy: public RtApi
-{
-public:
-
- RtApiDummy() { errorText_ = "RtApiDummy: This class provides no functionality."; error( RtAudioError::WARNING ); }
- RtAudio::Api getCurrentApi( void ) { return RtAudio::RTAUDIO_DUMMY; }
- unsigned int getDeviceCount( void ) { return 0; }
- RtAudio::DeviceInfo getDeviceInfo( unsigned int /*device*/ ) { RtAudio::DeviceInfo info; return info; }
- void closeStream( void ) {}
- void startStream( void ) {}
- void stopStream( void ) {}
- void abortStream( void ) {}
-
- private:
-
- 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; }
-};
-
-#endif
+inline void RtAudio :: setErrorCallback( RtAudioErrorCallback errorCallback ) { rtapi_->setErrorCallback( errorCallback ); }
+inline void RtAudio :: showWarnings( bool value ) { rtapi_->showWarnings( value ); }
#endif
diff --git a/src/modules/rtaudio/consumer_rtaudio.cpp b/src/modules/rtaudio/consumer_rtaudio.cpp
index 1fe3f2de..acc55730 100644
--- a/src/modules/rtaudio/consumer_rtaudio.cpp
+++ b/src/modules/rtaudio/consumer_rtaudio.cpp
@@ -1,6 +1,6 @@
/*
* consumer_rtaudio.c -- output through RtAudio audio wrapper
- * Copyright (C) 2011-2021 Meltytech, LLC
+ * Copyright (C) 2011-2023 Meltytech, LLC
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
@@ -28,6 +28,10 @@
#include <RtAudio.h>
#endif
+#if defined(RTAUDIO_VERSION_MAJOR) && RTAUDIO_VERSION_MAJOR >= 6
+#define RTAUDIO_VERSION_6
+#endif
+
static void consumer_refresh_cb(mlt_consumer sdl, mlt_consumer consumer, mlt_event_data);
static int rtaudio_callback(void *outputBuffer,
void *inputBuffer,
@@ -93,9 +97,9 @@ public:
mlt_consumer getConsumer() { return &consumer; }
RtAudioConsumer()
- : rt(NULL)
+ : rt(nullptr)
, device_id(-1)
- , queue(NULL)
+ , queue(nullptr)
, joined(0)
, running(0)
, audio_avail(0)
@@ -122,7 +126,7 @@ public:
if (rt && rt->isStreamOpen())
rt->closeStream();
delete rt;
- rt = NULL;
+ rt = nullptr;
}
bool create_rtaudio(RtAudio::Api api, int channels, int frequency)
@@ -145,22 +149,34 @@ public:
if (rt->getDeviceCount() < 1) {
mlt_log_warning(getConsumer(), "no audio devices found\n");
delete rt;
- rt = NULL;
+ rt = nullptr;
return false;
}
-#ifndef __LINUX_ALSA__
+#if defined(RTAUDIO_VERSION_6) || !defined(__LINUX_ALSA__)
device_id = rt->getDefaultOutputDevice();
#endif
if (resource && strcmp(resource, "") && strcmp(resource, "default")) {
// Get device ID by name
- unsigned int n = rt->getDeviceCount();
RtAudio::DeviceInfo info;
unsigned int i;
+#ifdef RTAUDIO_VERSION_6
+ auto ids = rt->getDeviceIds();
+ for (i = 0; i < ids.size(); i++) {
+ info = rt->getDeviceInfo(ids[i]);
+ mlt_log_verbose(nullptr, "RtAudio device %u = %s\n", ids[i], info.name.c_str());
+ if (info.name.find(resource) != std::string::npos) {
+ device_id = info.ID;
+ break;
+ }
+ }
+ if (i == ids.size())
+#else
+ unsigned int n = rt->getDeviceCount();
for (i = 0; i < n; i++) {
info = rt->getDeviceInfo(i);
- mlt_log_verbose(NULL, "RtAudio device %d = %s\n", i, info.name.c_str());
+ mlt_log_verbose(nullptr, "RtAudio device %d = %s\n", i, info.name.c_str());
if (info.probed && info.name == resource) {
device_id = i;
break;
@@ -168,7 +184,8 @@ public:
}
// Name selection failed, try arg as numeric
if (i == n)
- device_id = (int) strtol(resource, NULL, 0);
+#endif
+ device_id = (int) strtol(resource, nullptr, 0);
}
RtAudio::StreamParameters parameters;
@@ -182,6 +199,16 @@ public:
parameters.deviceId = 0;
}
if (resource) {
+#ifdef RTAUDIO_VERSION_6
+ auto ids = rt->getDeviceIds();
+ for (unsigned i = 0; i < ids.size(); i++) {
+ auto info = rt->getDeviceInfo(ids[i]);
+ if (info.name == resource) {
+ device_id = parameters.deviceId = info.ID;
+ break;
+ }
+ }
+#else
unsigned n = rt->getDeviceCount();
for (unsigned i = 0; i < n; i++) {
RtAudio::DeviceInfo info = rt->getDeviceInfo(i);
@@ -190,14 +217,34 @@ public:
break;
}
}
+#endif
}
+#ifdef RTAUDIO_VERSION_6
+ if (rt->isStreamOpen()) {
+ rt->closeStream();
+ }
+ if (rt->openStream(&parameters,
+ nullptr,
+ RTAUDIO_SINT16,
+ frequency,
+ &bufferFrames,
+ &rtaudio_callback,
+ this,
+ &options)
+ || rt->startStream()) {
+ mlt_log_info(getConsumer(), "%s\n", rt->getErrorText().c_str());
+ delete rt;
+ rt = nullptr;
+ return false;
+ }
+#else
try {
if (rt->isStreamOpen()) {
rt->closeStream();
}
rt->openStream(&parameters,
- NULL,
+ nullptr,
RTAUDIO_SINT16,
frequency,
&bufferFrames,
@@ -213,9 +260,10 @@ public:
#endif
mlt_log_info(getConsumer(), "%s\n", e.getMessage().c_str());
delete rt;
- rt = NULL;
+ rt = nullptr;
return false;
}
+#endif
mlt_log_info(getConsumer(),
"Opened RtAudio: %s\t%d\t%d\n",
rtaudio_api_str(rt->getCurrentApi()),
@@ -288,10 +336,10 @@ public:
mlt_properties_set_double(properties, "volume", 1.0);
// This is the initialisation of the consumer
- pthread_mutex_init(&audio_mutex, NULL);
- pthread_cond_init(&audio_cond, NULL);
- pthread_mutex_init(&video_mutex, NULL);
- pthread_cond_init(&video_cond, NULL);
+ pthread_mutex_init(&audio_mutex, nullptr);
+ pthread_cond_init(&audio_cond, nullptr);
+ pthread_mutex_init(&video_mutex, nullptr);
+ pthread_cond_init(&video_cond, nullptr);
// Default scaler (for now we'll use nearest)
mlt_properties_set(properties, "rescale", "nearest");
@@ -310,8 +358,8 @@ public:
joined = 1;
// Initialize the refresh handler
- pthread_cond_init(&refresh_cond, NULL);
- pthread_mutex_init(&refresh_mutex, NULL);
+ pthread_cond_init(&refresh_cond, nullptr);
+ pthread_mutex_init(&refresh_mutex, nullptr);
mlt_events_listen(properties, this, "property-changed", (mlt_listener) consumer_refresh_cb);
return true;
@@ -323,7 +371,7 @@ public:
stop();
running = 1;
joined = 0;
- pthread_create(&thread, NULL, consumer_thread_proxy, this);
+ pthread_create(&thread, nullptr, consumer_thread_proxy, this);
}
return 0;
@@ -342,7 +390,7 @@ public:
pthread_mutex_unlock(&refresh_mutex);
// Cleanup the main thread
- pthread_join(thread, NULL);
+ pthread_join(thread, nullptr);
// Unlatch the video thread
pthread_mutex_lock(&video_mutex);
@@ -355,6 +403,11 @@ public:
pthread_mutex_unlock(&audio_mutex);
if (rt && rt->isStreamOpen())
+#ifdef RTAUDIO_VERSION_6
+ if (rt->stopStream()) {
+ mlt_log_error(getConsumer(), "%s\n", rt->getErrorText().c_str());
+ }
+#else
try {
// Stop the stream
rt->stopStream();
@@ -366,8 +419,9 @@ public:
#endif
mlt_log_error(getConsumer(), "%s\n", e.getMessage().c_str());
}
+#endif
delete rt;
- rt = NULL;
+ rt = nullptr;
}
return 0;
@@ -402,8 +456,8 @@ public:
// internal initialization
int init_audio = 1;
int init_video = 1;
- mlt_frame frame = NULL;
- mlt_properties properties = NULL;
+ mlt_frame frame = nullptr;
+ mlt_properties properties = nullptr;
int64_t duration = 0;
int64_t playtime = mlt_properties_get_int(consumer_props, "video_delay") * 1000;
struct timespec tm = {0, 100000};
@@ -440,7 +494,7 @@ public:
// Determine the start time now
if (playing && init_video) {
// Create the video thread
- pthread_create(&thread, NULL, video_thread_proxy, this);
+ pthread_create(&thread, nullptr, video_thread_proxy, this);
// Video doesn't need to be initialised any more
init_video = 0;
@@ -450,7 +504,7 @@ public:
mlt_properties_set_int64(properties, "playtime", playtime);
while (running && speed != 0 && mlt_deque_count(queue) > 15)
- nanosleep(&tm, NULL);
+ nanosleep(&tm, nullptr);
// Push this frame to the back of the video queue
if (running && speed) {
@@ -477,7 +531,7 @@ public:
pthread_mutex_unlock(&refresh_mutex);
} else {
mlt_frame_close(frame);
- frame = NULL;
+ frame = nullptr;
}
// Optimisation to reduce latency
@@ -498,7 +552,7 @@ public:
pthread_mutex_lock(&video_mutex);
pthread_cond_broadcast(&video_cond);
pthread_mutex_unlock(&video_mutex);
- pthread_join(thread, NULL);
+ pthread_join(thread, nullptr);
}
while (mlt_deque_count(queue))
@@ -588,7 +642,7 @@ public:
init_audio = 0;
playing = 1;
} else {
- rt = NULL;
+ rt = nullptr;
mlt_log_error(getConsumer(), "Unable to initialize RtAudio\n");
init_audio = 2;
}
@@ -661,7 +715,7 @@ public:
int64_t start = 0;
int64_t elapsed = 0;
struct timespec tm;
- mlt_frame next = NULL;
+ mlt_frame next = nullptr;
mlt_properties consumerProperties = MLT_CONSUMER_PROPERTIES(getConsumer());
double speed = 0;
@@ -669,7 +723,7 @@ public:
int real_time = mlt_properties_get_int(consumerProperties, "real_time");
// Get the current time
- gettimeofday(&now, NULL);
+ gettimeofday(&now, nullptr);
// Determine start time
start = (int64_t) now.tv_sec * 1000000 + now.tv_usec;
@@ -678,13 +732,13 @@ public:
// Pop the next frame
pthread_mutex_lock(&video_mutex);
next = (mlt_frame) mlt_deque_pop_front(queue);
- while (next == NULL && running) {
+ while (next == nullptr && running) {
pthread_cond_wait(&video_cond, &video_mutex);
next = (mlt_frame) mlt_deque_pop_front(queue);
}
pthread_mutex_unlock(&video_mutex);
- if (!running || next == NULL)
+ if (!running || next == nullptr)
break;
// Get the properties
@@ -694,7 +748,7 @@ public:
speed = mlt_properties_get_double(properties, "_speed");
// Get the current time
- gettimeofday(&now, NULL);
+ gettimeofday(&now, nullptr);
// Get the elapsed time
elapsed = ((int64_t) now.tv_sec * 1000000 + now.tv_usec) - start;
@@ -711,7 +765,7 @@ public:
if (real_time && (difference > 20000 && speed == 1.0)) {
tm.tv_sec = difference / 1000000;
tm.tv_nsec = (difference % 1000000) * 1000;
- nanosleep(&tm, NULL);
+ nanosleep(&tm, nullptr);
}
// Show current frame if not too old
@@ -721,7 +775,7 @@ public:
// If the queue is empty, recalculate start to allow build up again
if (real_time && (mlt_deque_count(queue) == 0 && speed == 1.0)) {
- gettimeofday(&now, NULL);
+ gettimeofday(&now, nullptr);
start = ((int64_t) now.tv_sec * 1000000 + now.tv_usec) - scheduled + 20000;
start += mlt_properties_get_int(consumerProperties, "video_delay") * 1000;
}
@@ -729,10 +783,10 @@ public:
// This frame can now be closed
mlt_frame_close(next);
- next = NULL;
+ next = nullptr;
}
- if (next != NULL)
+ if (next != nullptr)
mlt_frame_close(next);
mlt_consumer_stopped(getConsumer());
@@ -770,14 +824,14 @@ static void *consumer_thread_proxy(void *arg)
{
RtAudioConsumer *rtaudio = (RtAudioConsumer *) arg;
rtaudio->consumer_thread();
- return NULL;
+ return nullptr;
}
static void *video_thread_proxy(void *arg)
{
RtAudioConsumer *rtaudio = (RtAudioConsumer *) arg;
rtaudio->video_thread();
- return NULL;
+ return nullptr;
}
/** Start the consumer.
@@ -822,7 +876,7 @@ static void close(mlt_consumer consumer)
mlt_consumer_stop(consumer);
// Close the parent
- consumer->close = NULL;
+ consumer->close = nullptr;
mlt_consumer_close(consumer);
// Free the memory
@@ -838,7 +892,7 @@ mlt_consumer consumer_rtaudio_init(mlt_profile profile,
{
// Allocate the consumer
RtAudioConsumer *rtaudio = new RtAudioConsumer();
- mlt_consumer consumer = NULL;
+ mlt_consumer consumer = nullptr;
// If allocated
if (rtaudio && !mlt_consumer_init(rtaudio->getConsumer(), rtaudio, profile)) {
@@ -872,7 +926,7 @@ static mlt_properties metadata(mlt_service_type type, const char *id, void *data
MLT_REPOSITORY
{
MLT_REGISTER(mlt_service_consumer_type, "rtaudio", consumer_rtaudio_init);
- MLT_REGISTER_METADATA(mlt_service_consumer_type, "rtaudio", metadata, NULL);
+ MLT_REGISTER_METADATA(mlt_service_consumer_type, "rtaudio", metadata, nullptr);
}
} // extern C
--
2.35.3
Reference in New Issue View Git Blame Copy Permalink