proteaaudio 0.8.0 → 0.9.0
raw patch · 16 files changed
+12040/−9058 lines, 16 filesPVP ok
version bump matches the API change (PVP)
API changes (from Hackage documentation)
+ Sound.ProteaAudio: sampleDestroy :: Sample -> IO Bool
+ Sound.ProteaAudio: soundActiveAll :: IO Int
+ Sound.ProteaAudio: soundLoopOn :: Int -> Sample -> Float -> Float -> Float -> Float -> IO Sound
+ Sound.ProteaAudio: soundPlayOn :: Int -> Sample -> Float -> Float -> Float -> Float -> IO Sound
- Sound.ProteaAudio: soundActive :: IO Int
+ Sound.ProteaAudio: soundActive :: Sound -> IO Bool
Files
- CHANGELOG.md +11/−0
- README.md +42/−29
- Sound/ProteaAudio.chs +50/−9
- cbits/RtAudio.cpp +10834/−8301
- cbits/RtAudio.h +273/−85
- cbits/RtError.h +0/−60
- cbits/proAudio.cpp +107/−107
- cbits/proAudio.h +118/−76
- cbits/proAudioRt.cpp +144/−119
- cbits/proAudioRt.h +45/−25
- cbits/proteaaudio_binding.cpp +59/−11
- cbits/proteaaudio_binding.h +9/−3
- cbits/stb_vorbis.c +295/−202
- example/play.hs +34/−28
- proteaaudio.cabal +6/−3
- stack.yaml +13/−0
CHANGELOG.md view
@@ -1,3 +1,14 @@+# 0.9.0+- rename `soundActive` to `soundActiveAll`+- `soundActive` now queries a specific sound++# 0.8.1+- update RtAudio to 5.1.0+- update stb_vorbis to v1.19+- add sampleDestroy+- add soundPlayOn and soundLoopOn+- fix: generate unique handles for Sample and Sound types+ # 0.8.0 - fix: distinct data and playback handles - introduce Sound handle type, an abstraction for playback audio track
README.md view
@@ -3,7 +3,7 @@ ProteaAudio is a stereo audio mixer/playback library for - Linux *(PulseAudio)*-- Macintosh OS X *(CoreAudio)*+- MacOS *(CoreAudio)* - Windows *(DirectSound)* Supported audio formats:@@ -13,6 +13,13 @@ Samples can be loaded from file or memory. +# Setup++On Linux you need to install the pulse audio library:+```+sudo apt install libpulse-dev+```+ # Build ### Stack@@ -39,45 +46,51 @@ import Sound.ProteaAudio -waitPayback = do- n <- soundActive+waitPlayback = do+ n <- soundActiveAll when (n > 0) $ do threadDelay oneSec- waitPayback+ waitPlayback oneSec :: Int oneSec = 1000000 -- micro seconds main = do- args <- getArgs- filename <- case args of- a : _ -> pure a- _ -> fail "usage: proteaaudio-play SAMPLE_FILE_NAME"+ args <- getArgs+ filename <- case args of+ a : _ -> pure a+ _ -> fail "usage: proteaaudio-play SAMPLE_FILE_NAME" - result <- initAudio 64 44100 1024 -- max channels, mixing frequency, mixing buffer size- unless result $ fail "failed to initialize the audio system"+ result <- initAudio 64 44100 1024 -- max channels, mixing frequency, mixing buffer size+ unless result $ fail "failed to initialize the audio system" - -- (A) load sample from file- sampleA <- sampleFromFile filename 1.0 -- volume+ -- (A) load sample from file+ sampleA <- sampleFromFile filename 1.0 -- volume - -- start two sound tracks with shared sample data- sndTrkA <- soundPlay sampleA 1 1 0 1 -- left volume, right volume, time difference between left and right, pitch factor for playback- threadDelay oneSec -- wait 1 sec- sndTrkB <- soundPlay sampleA 1 1 0 1 -- left volume, right volume, time difference between left and right, pitch factor for playback- -- play 3 sec- threadDelay $ 3 * oneSec- soundStop sndTrkB- -- wait sndTrkA to finish- waitPayback+ -- start two sound tracks with shared sample data+ sndTrkA <- soundPlay sampleA 1 1 0 1 -- left volume, right volume, time difference between left and right, pitch factor for playback+ threadDelay oneSec -- wait 1 sec+ sndTrkB <- soundPlay sampleA 1 1 0 1 -- left volume, right volume, time difference between left and right, pitch factor for playback+ soundActive sndTrkB >>= print+ -- play 3 sec+ threadDelay $ 3 * oneSec+ soundStop sndTrkB+ soundActive sndTrkB >>= print+ -- wait sndTrkA to finish+ waitPlayback - -- (B) load from memory buffer- buffer <- SB.readFile filename- sampleB <- case takeExtension filename of- ".ogg" -> sampleFromMemoryOgg buffer 1.0- ".wav" -> sampleFromMemoryWav buffer 1.0+ -- (B) load from memory buffer+ buffer <- SB.readFile filename+ sampleB <- case takeExtension filename of+ ".ogg" -> sampleFromMemoryOgg buffer 1.0+ ".wav" -> sampleFromMemoryWav buffer 1.0 - soundPlay sampleB 1 1 0 1 -- left volume, right volume, time difference between left and right, pitch factor for playback- waitPayback+ soundPlay sampleB 1 1 0 1 -- left volume, right volume, time difference between left and right, pitch factor for playback+ waitPlayback - finishAudio+ sampleDestroy sampleB+ soundPlay sampleB 1 1 0 1 -- we have invalidated the handle; nothing should happen now+ waitPlayback++ finishAudio ```
Sound/ProteaAudio.chs view
@@ -1,4 +1,13 @@+{-#LANGUAGE ForeignFunctionInterface,CPP#-}+#include "proteaaudio_binding.h"++#if defined PROTEAAUDIO_SDL {-|+ProteaAudio-SDL is a stereo audio mixer and playback library for SDL /(platform independent)/+-}+module Sound.ProteaAudio.SDL (+#elif defined PROTEAAUDIO_RT+{-| ProteaAudio is a stereo audio mixer/playback library for * Linux /(PusleAudio)/@@ -7,9 +16,10 @@ * Windows /(DirectSound)/ -}-{-#LANGUAGE ForeignFunctionInterface#-}-#include "proteaaudio_binding.h" module Sound.ProteaAudio (+#else+#error "I can't decide which binding to compile; please define either PROTEAAUDIO_SDL or PROTEAAUDIO_RT"+#endif -- * Sample Sample(), Sound(),@@ -20,7 +30,7 @@ -- * Main Mixer volume,- soundActive,+ soundActiveAll, soundStopAll, -- * Sample Loading@@ -29,12 +39,16 @@ sampleFromMemoryWav, sampleFromMemoryOgg, sampleFromFile,+ sampleDestroy, -- * Sample Playback soundLoop, soundPlay,+ soundLoopOn,+ soundPlayOn, soundUpdate,- soundStop+ soundStop,+ soundActive ) where import Foreign@@ -50,7 +64,7 @@ -- | Initializes the audio system. {#fun initAudio- { `Int' -- ^ the maximum number of sounds that are played parallely. Computation time is linearly correlated to this factor.+ { `Int' -- ^ the maximum number of sounds that are played parallely, at most 1024. Computation time is linearly correlated to this factor. , `Int' -- ^ sample frequency of the playback in Hz. 22050 corresponds to FM radio 44100 is CD quality. Computation time is linearly correlated to this factor. , `Int' -- ^ the number of bytes that are sent to the sound card at once. Low numbers lead to smaller latencies but need more computation time (thread switches). If a too small number is chosen, the sounds might not be played continuously. The default value 512 guarantees a good latency below 40 ms at 22050 Hz sample frequency. } -> `Bool' -- ^ returns True on success@@ -124,6 +138,9 @@ } -> `Sample' Sample -- ^ returns handle #} +-- | Unloads a previously loaded sample from memory, invalidating the handle.+{#fun sampleDestroy {fromSample `Sample'} -> `Bool'#}+ -- | Set main mixer volume. {#fun volume { `Float' -- ^ left@@ -132,12 +149,12 @@ #} -- | Return the number of currently active sounds.-{#fun soundActive {} -> `Int'#}+{#fun soundActiveAll {} -> `Int'#} -- | Stops all sounds immediately. {#fun soundStopAll {} -> `()'#} --- | Plays a specified sound sample continuously and sets its parameters.+-- | Plays a specified sound sample continuously any free channel and sets its parameters. {#fun soundLoop { fromSample `Sample' -- ^ handle of a previously loaded sample , `Float' -- ^ left volume@@ -147,7 +164,7 @@ } -> `Sound' Sound #} --- | Plays a specified sound sample once and sets its parameters.+-- | Plays a specified sound sample once any free channel and sets its parameters. {#fun soundPlay { fromSample `Sample' -- ^ handle of a previously loaded sample , `Float' -- ^ left volume@@ -157,9 +174,31 @@ } -> `Sound' Sound #} +-- | Plays a specified sound sample once on a specific channel and sets its parameters.+{#fun soundPlayOn+ { `Int' -- ^ number of the channel to use for playback with the first channel starting at 0+ , fromSample `Sample' -- ^ handle of a previously loaded sample+ , `Float' -- ^ left volume+ , `Float' -- ^ right volume+ , `Float' -- ^ time difference between left and right channel in seconds. Use negative values to specify a delay for the left channel, positive for the right+ , `Float' -- ^ pitch factor for playback. 0.5 corresponds to one octave below, 2.0 to one above the original sample+ } -> `Sound' Sound+#}++-- | Plays a specified sound sample continuously on a specific channel and sets its parameters.+{#fun soundLoopOn+ { `Int' -- ^ number of the channel to use for playback with the first channel starting at 0+ , fromSample `Sample' -- ^ handle of a previously loaded sample+ , `Float' -- ^ left volume+ , `Float' -- ^ right volume+ , `Float' -- ^ time difference between left and right channel in seconds. Use negative values to specify a delay for the left channel, positive for the right+ , `Float' -- ^ pitch factor for playback. 0.5 corresponds to one octave below, 2.0 to one above the original sample+ } -> `Sound' Sound+#}+ -- | Updates parameters of a specified sound. {#fun soundUpdate- { fromSound `Sound' -- ^ handle of a currently active sound+ { fromSound `Sound' -- ^ handle of a currently active sound (if sound has stopped, this is a no-op) , `Float' -- ^ left volume , `Float' -- ^ right volume , `Float' -- ^ time difference between left and right channel in seconds. Use negative values to specify a delay for the left channel, positive for the right@@ -170,3 +209,5 @@ -- | Stops a specified sound immediately. {#fun soundStop {fromSound `Sound'} -> `Bool'#} +-- | Checks if a specified sound is still active.+{#fun soundActive{fromSound `Sound'} -> `Bool'#}
cbits/RtAudio.cpp view
@@ -5,8307 +5,10840 @@ 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 and ASIO) operating systems. - - RtAudio WWW site: http://www.music.mcgill.ca/~gary/rtaudio/ - - RtAudio: realtime audio i/o C++ classes - Copyright (c) 2001-2012 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 4.0.11 - -#include "RtAudio.h" -#include <iostream> -#include <cstdlib> -#include <cstring> -#include <climits> - -// 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__) - #define MUTEX_INITIALIZE(A) InitializeCriticalSection(A) - #define MUTEX_DESTROY(A) DeleteCriticalSection(A) - #define MUTEX_LOCK(A) EnterCriticalSection(A) - #define MUTEX_UNLOCK(A) LeaveCriticalSection(A) -#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. -// -// *************************************************** // - -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_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_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 ) throw() -{ - 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_->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 print out an error message. - std::cerr << "\nRtAudio: no compiled API support found ... critical error!!\n\n"; -} - -RtAudio :: ~RtAudio() throw() -{ - 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 ) -{ - 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() -{ - stream_.state = STREAM_CLOSED; - stream_.mode = UNINITIALIZED; - stream_.apiHandle = 0; - stream_.userBuffer[0] = 0; - stream_.userBuffer[1] = 0; - MUTEX_INITIALIZE( &stream_.mutex ); - showWarnings_ = true; -} - -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 ) -{ - if ( stream_.state != STREAM_CLOSED ) { - errorText_ = "RtApi::openStream: a stream is already open!"; - error( RtError::INVALID_USE ); - } - - if ( oParams && oParams->nChannels < 1 ) { - errorText_ = "RtApi::openStream: a non-NULL output StreamParameters structure cannot have an nChannels value less than one."; - error( RtError::INVALID_USE ); - } - - if ( iParams && iParams->nChannels < 1 ) { - errorText_ = "RtApi::openStream: a non-NULL input StreamParameters structure cannot have an nChannels value less than one."; - error( RtError::INVALID_USE ); - } - - if ( oParams == NULL && iParams == NULL ) { - errorText_ = "RtApi::openStream: input and output StreamParameters structures are both NULL!"; - error( RtError::INVALID_USE ); - } - - if ( formatBytes(format) == 0 ) { - errorText_ = "RtApi::openStream: 'format' parameter value is undefined."; - error( RtError::INVALID_USE ); - } - - 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( RtError::INVALID_USE ); - } - } - - unsigned int iChannels = 0; - if ( iParams ) { - iChannels = iParams->nChannels; - if ( iParams->deviceId >= nDevices ) { - errorText_ = "RtApi::openStream: input device parameter value is invalid."; - error( RtError::INVALID_USE ); - } - } - - clearStreamInfo(); - bool result; - - if ( oChannels > 0 ) { - - result = probeDeviceOpen( oParams->deviceId, OUTPUT, oChannels, oParams->firstChannel, - sampleRate, format, bufferFrames, options ); - if ( result == false ) error( RtError::SYSTEM_ERROR ); - } - - if ( iChannels > 0 ) { - - result = probeDeviceOpen( iParams->deviceId, INPUT, iChannels, iParams->firstChannel, - sampleRate, format, bufferFrames, options ); - if ( result == false ) { - if ( oChannels > 0 ) closeStream(); - error( RtError::SYSTEM_ERROR ); - } - } - - stream_.callbackInfo.callback = (void *) callback; - stream_.callbackInfo.userData = userData; - - 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 -} - -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; } -}; - -ThreadHandle threadId; - -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( RtError::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( RtError::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( RtError::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( RtError::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( RtError::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( RtError::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( RtError::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( RtError::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( RtError::INVALID_USE ); - } - - if ( device >= nDevices ) { - errorText_ = "RtApiCore::getDeviceInfo: device ID is invalid!"; - error( RtError::INVALID_USE ); - } - - 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( RtError::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( RtError::WARNING ); - return info; - } - - //const char *mname = CFStringGetCStringPtr( cfname, CFStringGetSystemEncoding() ); - int length = CFStringGetLength(cfname); - char *mname = (char *)malloc(length * 3 + 1); - CFStringGetCString(cfname, mname, length * 3 + 1, CFStringGetSystemEncoding()); - 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( RtError::WARNING ); - return info; - } - - //const char *name = CFStringGetCStringPtr( cfname, CFStringGetSystemEncoding() ); - length = CFStringGetLength(cfname); - char *name = (char *)malloc(length * 3 + 1); - CFStringGetCString(cfname, name, length * 3 + 1, CFStringGetSystemEncoding()); - 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( RtError::WARNING ); - return info; - } - - // Allocate the AudioBufferList. - bufferList = (AudioBufferList *) malloc( dataSize ); - if ( bufferList == NULL ) { - errorText_ = "RtApiCore::getDeviceInfo: memory error allocating output AudioBufferList."; - error( RtError::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( RtError::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( RtError::WARNING ); - return info; - } - - // Allocate the AudioBufferList. - bufferList = (AudioBufferList *) malloc( dataSize ); - if ( bufferList == NULL ) { - errorText_ = "RtApiCore::getDeviceInfo: memory error allocating input AudioBufferList."; - error( RtError::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( RtError::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( RtError::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( RtError::WARNING ); - return info; - } - - Float64 minimumRate = 100000000.0, maximumRate = 0.0; - for ( UInt32 i=0; i<nRanges; i++ ) { - if ( rangeList[i].mMinimum < minimumRate ) minimumRate = rangeList[i].mMinimum; - if ( rangeList[i].mMaximum > maximumRate ) maximumRate = rangeList[i].mMaximum; - } - - info.sampleRates.clear(); - 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.sampleRates.size() == 0 ) { - errorStream_ << "RtApiCore::probeDeviceInfo: No supported sample rates found for device (" << device << ")."; - errorText_ = errorStream_.str(); - error( RtError::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; -} - -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; -} - -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; -} - -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) { - 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 ) { - 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( RtError::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; - 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; - } - - return FAILURE; -} - -void RtApiCore :: closeStream( void ) -{ - if ( stream_.state == STREAM_CLOSED ) { - errorText_ = "RtApiCore::closeStream(): no open stream to close!"; - error( RtError::WARNING ); - return; - } - - CoreHandle *handle = (CoreHandle *) stream_.apiHandle; - if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { - 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 ( 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( RtError::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( RtError::SYSTEM_ERROR ); -} - -void RtApiCore :: stopStream( void ) -{ - verifyStream(); - if ( stream_.state == STREAM_STOPPED ) { - errorText_ = "RtApiCore::stopStream(): the stream is already stopped!"; - error( RtError::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( RtError::SYSTEM_ERROR ); -} - -void RtApiCore :: abortStream( void ) -{ - verifyStream(); - if ( stream_.state == STREAM_STOPPED ) { - errorText_ = "RtApiCore::abortStream(): the stream is already stopped!"; - error( RtError::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. -extern "C" 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( RtError::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 ) { - - 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; - } - } - } - - 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; } -}; - -ThreadHandle threadId; -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( RtError::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( RtError::INVALID_USE ); - } - - // Get the current jack server sample rate. - info.sampleRates.clear(); - info.sampleRates.push_back( jack_get_sample_rate( client ) ); - - // 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( RtError::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; -} - -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. -extern "C" void *jackCloseStream( void *ptr ) -{ - CallbackInfo *info = (CallbackInfo *) ptr; - RtApiJack *object = (RtApiJack *) info->object; - - object->closeStream(); - - pthread_exit( NULL ); -} -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; - - pthread_create( &threadId, NULL, jackCloseStream, info ); - std::cerr << "\nRtApiJack: the Jack server is shutting down this client ... stream stopped and closed!!\n" << std::endl; -} - -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( RtError::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 ] ) - 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( RtError::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( RtError::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( RtError::SYSTEM_ERROR ); -} - -void RtApiJack :: stopStream( void ) -{ - verifyStream(); - if ( stream_.state == STREAM_STOPPED ) { - errorText_ = "RtApiJack::stopStream(): the stream is already stopped!"; - error( RtError::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( RtError::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. -extern "C" 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( RtError::WARNING ); - return FAILURE; - } - if ( stream_.bufferSize != nframes ) { - errorText_ = "RtApiCore::callbackEvent(): the JACK buffer size has changed ... cannot process!"; - error( RtError::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 ) { - - 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 ); - } - } - - 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> - -AsioDrivers drivers; -ASIOCallbacks asioCallbacks; -ASIODriverInfo driverInfo; -CallbackInfo *asioCallbackInfo; -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 ); -void sampleRateChanged( ASIOSampleRate sRate ); -long asioMessages( long selector, long value, void* message, double* opt ); - -RtApiAsio :: RtApiAsio() -{ - // ASIO cannot run on a multi-threaded appartment. You can call - // CoInitialize beforehand, but it must be for appartment threading - // (in which case, CoInitilialize will return S_FALSE here). - coInitialized_ = false; - HRESULT hr = CoInitialize( NULL ); - if ( FAILED(hr) ) { - errorText_ = "RtApiAsio::ASIO requires a single-threaded appartment. Call CoInitializeEx(0,COINIT_APARTMENTTHREADED)"; - error( RtError::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( RtError::INVALID_USE ); - } - - if ( device >= nDevices ) { - errorText_ = "RtApiAsio::getDeviceInfo: device ID is invalid!"; - error( RtError::INVALID_USE ); - } - - // 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( RtError::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( RtError::WARNING ); - return info; - } - - info.name = driverName; - - if ( !drivers.loadDriver( driverName ) ) { - errorStream_ << "RtApiAsio::getDeviceInfo: unable to load driver (" << driverName << ")."; - errorText_ = errorStream_.str(); - error( RtError::WARNING ); - return info; - } - - result = ASIOInit( &driverInfo ); - if ( result != ASE_OK ) { - errorStream_ << "RtApiAsio::getDeviceInfo: error (" << getAsioErrorString( result ) << ") initializing driver (" << driverName << ")."; - errorText_ = errorStream_.str(); - error( RtError::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( RtError::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] ); - } - - // 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( RtError::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; - - 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; -} - -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 ) -{ - // For ASIO, a duplex stream MUST use the same driver. - if ( mode == INPUT && stream_.mode == OUTPUT && 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 ( mode != INPUT || stream_.mode != OUTPUT ) { - // 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; - } - } - - // Check the device channel count. - long inputChannels, outputChannels; - result = ASIOGetChannels( &inputChannels, &outputChannels ); - if ( result != ASE_OK ) { - drivers.removeCurrentDriver(); - errorStream_ << "RtApiAsio::probeDeviceOpen: error (" << getAsioErrorString( result ) << ") getting channel count (" << driverName << ")."; - errorText_ = errorStream_.str(); - return FAILURE; - } - - if ( ( mode == OUTPUT && (channels+firstChannel) > (unsigned int) outputChannels) || - ( mode == INPUT && (channels+firstChannel) > (unsigned int) inputChannels) ) { - drivers.removeCurrentDriver(); - errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") does not support requested channel count (" << channels << ") + offset (" << firstChannel << ")."; - errorText_ = errorStream_.str(); - return FAILURE; - } - 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 ) { - drivers.removeCurrentDriver(); - errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") does not support requested sample rate (" << sampleRate << ")."; - errorText_ = errorStream_.str(); - return FAILURE; - } - - // Get the current sample rate - ASIOSampleRate currentRate; - result = ASIOGetSampleRate( ¤tRate ); - if ( result != ASE_OK ) { - drivers.removeCurrentDriver(); - errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error getting sample rate."; - errorText_ = errorStream_.str(); - return FAILURE; - } - - // Set the sample rate only if necessary - if ( currentRate != sampleRate ) { - result = ASIOSetSampleRate( (ASIOSampleRate) sampleRate ); - if ( result != ASE_OK ) { - drivers.removeCurrentDriver(); - errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error setting sample rate (" << sampleRate << ")."; - errorText_ = errorStream_.str(); - return FAILURE; - } - } - - // 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 ) { - drivers.removeCurrentDriver(); - errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error (" << getAsioErrorString( result ) << ") getting data format."; - errorText_ = errorStream_.str(); - return FAILURE; - } - - // 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; - } - - if ( stream_.deviceFormat[mode] == 0 ) { - drivers.removeCurrentDriver(); - errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") data format not supported by RtAudio."; - errorText_ = errorStream_.str(); - return FAILURE; - } - - // 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 ) { - drivers.removeCurrentDriver(); - errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error (" << getAsioErrorString( result ) << ") getting buffer size."; - errorText_ = errorStream_.str(); - return FAILURE; - } - - 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; - } - - if ( mode == INPUT && stream_.mode == OUTPUT && stream_.bufferSize != *bufferSize ) { - drivers.removeCurrentDriver(); - errorText_ = "RtApiAsio::probeDeviceOpen: input/output buffersize discrepancy!"; - return FAILURE; - } - - 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. - AsioHandle *handle = (AsioHandle *) stream_.apiHandle; - if ( handle == 0 ) { - try { - handle = new AsioHandle; - } - catch ( std::bad_alloc& ) { - //if ( handle == NULL ) { - drivers.removeCurrentDriver(); - errorText_ = "RtApiAsio::probeDeviceOpen: error allocating AsioHandle memory."; - return FAILURE; - } - 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. - long inputLatency, outputLatency; - if ( mode == INPUT && stream_.mode == OUTPUT ) { - ASIODisposeBuffers(); - if ( handle->bufferInfos ) free( handle->bufferInfos ); - } - - // Allocate, initialize, and save the bufferInfos in our stream callbackInfo structure. - bool buffersAllocated = false; - 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; - } - - // 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 ) { - errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error (" << getAsioErrorString( result ) << ") creating buffers."; - errorText_ = errorStream_.str(); - goto error; - } - buffersAllocated = true; - - // 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 ( 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_ = "RtApiAsio::probeDeviceOpen: error allocating device buffer memory."; - goto error; - } - } - } - - stream_.sampleRate = sampleRate; - stream_.device[mode] = device; - stream_.state = STREAM_STOPPED; - asioCallbackInfo = &stream_.callbackInfo; - stream_.callbackInfo.object = (void *) this; - if ( stream_.mode == OUTPUT && mode == INPUT ) - // We had already set up an output stream. - stream_.mode = DUPLEX; - else - stream_.mode = mode; - - // Determine device latencies - result = ASIOGetLatencies( &inputLatency, &outputLatency ); - if ( result != ASE_OK ) { - errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error (" << getAsioErrorString( result ) << ") getting latency."; - errorText_ = errorStream_.str(); - error( RtError::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 ( buffersAllocated ) - ASIODisposeBuffers(); - drivers.removeCurrentDriver(); - - 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; - } - - return FAILURE; -} - -void RtApiAsio :: closeStream() -{ - if ( stream_.state == STREAM_CLOSED ) { - errorText_ = "RtApiAsio::closeStream(): no open stream to close!"; - error( RtError::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( RtError::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( RtError::SYSTEM_ERROR ); -} - -void RtApiAsio :: stopStream() -{ - verifyStream(); - if ( stream_.state == STREAM_STOPPED ) { - errorText_ = "RtApiAsio::stopStream(): the stream is already stopped!"; - error( RtError::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( RtError::SYSTEM_ERROR ); -} - -void RtApiAsio :: abortStream() -{ - verifyStream(); - if ( stream_.state == STREAM_STOPPED ) { - errorText_ = "RtApiAsio::abortStream(): the stream is already stopped!"; - error( RtError::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. -extern "C" 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( RtError::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 ); - } - - } - - 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; -} - -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 ( RtError &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; -} - -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 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_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 ); - -extern "C" 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; } -}; - -std::vector< 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. - bool isInput = false; - HRESULT result = DirectSoundEnumerate( (LPDSENUMCALLBACK) deviceQueryCallback, &isInput ); - if ( FAILED( result ) ) { - errorStream_ << "RtApiDs::getDeviceCount: error (" << getErrorString( result ) << ") enumerating output devices!"; - errorText_ = errorStream_.str(); - error( RtError::WARNING ); - } - - // Query DirectSoundCapture devices. - isInput = true; - result = DirectSoundCaptureEnumerate( (LPDSENUMCALLBACK) deviceQueryCallback, &isInput ); - if ( FAILED( result ) ) { - errorStream_ << "RtApiDs::getDeviceCount: error (" << getErrorString( result ) << ") enumerating input devices!"; - errorText_ = errorStream_.str(); - error( RtError::WARNING ); - } - - // Clean out any devices that may have disappeared. - std::vector< int > indices; - for ( unsigned int i=0; i<dsDevices.size(); i++ ) - if ( dsDevices[i].found == false ) indices.push_back( i ); - unsigned int nErased = 0; - for ( unsigned int i=0; i<indices.size(); i++ ) - dsDevices.erase( dsDevices.begin()-nErased++ ); - - return 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( RtError::INVALID_USE ); - } - } - - if ( device >= dsDevices.size() ) { - errorText_ = "RtApiDs::getDeviceInfo: device ID is invalid!"; - error( RtError::INVALID_USE ); - } - - 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( RtError::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( RtError::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] ); - } - - // 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( RtError::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( RtError::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; - } - - unsigned int 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; - } - - return FAILURE; -} - -void RtApiDs :: closeStream() -{ - if ( stream_.state == STREAM_CLOSED ) { - errorText_ = "RtApiDs::closeStream(): no open stream to close!"; - error( RtError::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( RtError::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( RtError::SYSTEM_ERROR ); -} - -void RtApiDs :: stopStream() -{ - verifyStream(); - if ( stream_.state == STREAM_STOPPED ) { - errorText_ = "RtApiDs::stopStream(): the stream is already stopped!"; - error( RtError::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; - - // 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; - - 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. - if ( FAILED( result ) ) error( RtError::SYSTEM_ERROR ); -} - -void RtApiDs :: abortStream() -{ - verifyStream(); - if ( stream_.state == STREAM_STOPPED ) { - errorText_ = "RtApiDs::abortStream(): the stream is already stopped!"; - error( RtError::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( RtError::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; - - 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(); - error( RtError::SYSTEM_ERROR ); - } - result = dsCaptureBuffer->GetCurrentPosition( NULL, &startSafeReadPointer ); - if ( FAILED( result ) ) { - errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current read position!"; - errorText_ = errorStream_.str(); - error( RtError::SYSTEM_ERROR ); - } - while ( true ) { - result = dsWriteBuffer->GetCurrentPosition( NULL, &safeWritePointer ); - if ( FAILED( result ) ) { - errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current write position!"; - errorText_ = errorStream_.str(); - error( RtError::SYSTEM_ERROR ); - } - result = dsCaptureBuffer->GetCurrentPosition( NULL, &safeReadPointer ); - if ( FAILED( result ) ) { - errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current read position!"; - errorText_ = errorStream_.str(); - error( RtError::SYSTEM_ERROR ); - } - if ( safeWritePointer != startSafeWritePointer && safeReadPointer != startSafeReadPointer ) break; - Sleep( 1 ); - } - - //assert( handle->dsBufferSize[0] == handle->dsBufferSize[1] ); - - handle->bufferPointer[0] = safeWritePointer + handle->dsPointerLeadTime[0]; - if ( handle->bufferPointer[0] >= handle->dsBufferSize[0] ) handle->bufferPointer[0] -= handle->dsBufferSize[0]; - handle->bufferPointer[1] = safeReadPointer; - } - else if ( stream_.mode == OUTPUT ) { - - // Set the proper nextWritePosition after initial startup. - LPDIRECTSOUNDBUFFER dsWriteBuffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0]; - result = dsWriteBuffer->GetCurrentPosition( ¤tWritePointer, &safeWritePointer ); - if ( FAILED( result ) ) { - errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current write position!"; - errorText_ = errorStream_.str(); - error( RtError::SYSTEM_ERROR ); - } - handle->bufferPointer[0] = safeWritePointer + handle->dsPointerLeadTime[0]; - if ( handle->bufferPointer[0] >= handle->dsBufferSize[0] ) handle->bufferPointer[0] -= handle->dsBufferSize[0]; - } - - buffersRolling = true; - } - - if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { - - LPDIRECTSOUNDBUFFER dsBuffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0]; - - if ( handle->drainCounter > 1 ) { // write zeros to the output stream - bufferBytes = stream_.bufferSize * stream_.nUserChannels[0]; - bufferBytes *= formatBytes( stream_.userFormat ); - memset( stream_.userBuffer[0], 0, bufferBytes ); - } - - // Setup parameters and do buffer conversion if necessary. - if ( stream_.doConvertBuffer[0] ) { - buffer = stream_.deviceBuffer; - convertBuffer( buffer, stream_.userBuffer[0], stream_.convertInfo[0] ); - bufferBytes = stream_.bufferSize * stream_.nDeviceChannels[0]; - bufferBytes *= formatBytes( stream_.deviceFormat[0] ); - } - else { - buffer = stream_.userBuffer[0]; - bufferBytes = stream_.bufferSize * stream_.nUserChannels[0]; - bufferBytes *= formatBytes( stream_.userFormat ); - } - - // No byte swapping necessary in DirectSound implementation. - - // Ahhh ... windoze. 16-bit data is signed but 8-bit data is - // unsigned. So, we need to convert our signed 8-bit data here to - // unsigned. - if ( stream_.deviceFormat[0] == RTAUDIO_SINT8 ) - for ( int i=0; i<bufferBytes; i++ ) buffer[i] = (unsigned char) ( buffer[i] + 128 ); - - DWORD dsBufferSize = handle->dsBufferSize[0]; - nextWritePointer = handle->bufferPointer[0]; - - DWORD endWrite, leadPointer; - while ( true ) { - // Find out where the read and "safe write" pointers are. - result = dsBuffer->GetCurrentPosition( ¤tWritePointer, &safeWritePointer ); - if ( FAILED( result ) ) { - errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current write position!"; - errorText_ = errorStream_.str(); - error( RtError::SYSTEM_ERROR ); - } - - // 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(); - error( RtError::SYSTEM_ERROR ); - } - - // 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(); - error( RtError::SYSTEM_ERROR ); - } - nextWritePointer = ( nextWritePointer + bufferSize1 + bufferSize2 ) % dsBufferSize; - handle->bufferPointer[0] = nextWritePointer; - - if ( handle->drainCounter ) { - handle->drainCounter++; - goto unlock; - } - } - - if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) { - - // Setup parameters. - if ( stream_.doConvertBuffer[1] ) { - buffer = stream_.deviceBuffer; - bufferBytes = stream_.bufferSize * stream_.nDeviceChannels[1]; - bufferBytes *= formatBytes( stream_.deviceFormat[1] ); - } - else { - buffer = stream_.userBuffer[1]; - bufferBytes = stream_.bufferSize * stream_.nUserChannels[1]; - bufferBytes *= formatBytes( stream_.userFormat ); - } - - LPDIRECTSOUNDCAPTUREBUFFER dsBuffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1]; - long nextReadPointer = handle->bufferPointer[1]; - DWORD dsBufferSize = handle->dsBufferSize[1]; - - // Find out where the write and "safe read" pointers are. - result = dsBuffer->GetCurrentPosition( ¤tReadPointer, &safeReadPointer ); - if ( FAILED( result ) ) { - errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current read position!"; - errorText_ = errorStream_.str(); - error( RtError::SYSTEM_ERROR ); - } - - if ( safeReadPointer < (DWORD)nextReadPointer ) safeReadPointer += dsBufferSize; // unwrap offset - DWORD endRead = nextReadPointer + bufferBytes; - - // Handling depends on whether we are INPUT or DUPLEX. - // If we're in INPUT mode then waiting is a good thing. If we're in DUPLEX mode, - // then a wait here will drag the write pointers into the forbidden zone. - // - // In DUPLEX mode, rather than wait, we will back off the read pointer until - // it's in a safe position. This causes dropouts, but it seems to be the only - // practical way to sync up the read and write pointers reliably, given the - // the very complex relationship between phase and increment of the read and write - // pointers. - // - // In order to minimize audible dropouts in DUPLEX mode, we will - // provide a pre-roll period of 0.5 seconds in which we return - // zeros from the read buffer while the pointers sync up. - - if ( stream_.mode == DUPLEX ) { - if ( safeReadPointer < endRead ) { - if ( duplexPrerollBytes <= 0 ) { - // Pre-roll time over. Be more agressive. - int adjustment = endRead-safeReadPointer; - - handle->xrun[1] = true; - // Two cases: - // - large adjustments: we've probably run out of CPU cycles, so just resync exactly, - // and perform fine adjustments later. - // - small adjustments: back off by twice as much. - if ( adjustment >= 2*bufferBytes ) - nextReadPointer = safeReadPointer-2*bufferBytes; - else - nextReadPointer = safeReadPointer-bufferBytes-adjustment; - - if ( nextReadPointer < 0 ) nextReadPointer += dsBufferSize; - - } - else { - // In pre=roll time. Just do it. - nextReadPointer = safeReadPointer - bufferBytes; - while ( nextReadPointer < 0 ) nextReadPointer += dsBufferSize; - } - endRead = nextReadPointer + bufferBytes; - } - } - else { // mode == INPUT - while ( safeReadPointer < endRead && stream_.callbackInfo.isRunning ) { - // See comments for playback. - double millis = (endRead - safeReadPointer) * 1000.0; - millis /= ( formatBytes(stream_.deviceFormat[1]) * stream_.nDeviceChannels[1] * stream_.sampleRate); - if ( millis < 1.0 ) millis = 1.0; - Sleep( (DWORD) millis ); - - // Wake up and find out where we are now. - result = dsBuffer->GetCurrentPosition( ¤tReadPointer, &safeReadPointer ); - if ( FAILED( result ) ) { - errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current read position!"; - errorText_ = errorStream_.str(); - error( RtError::SYSTEM_ERROR ); - } - - 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(); - error( RtError::SYSTEM_ERROR ); - } - - 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(); - error( RtError::SYSTEM_ERROR ); - } - 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: - RtApi::tickStreamTime(); -} - -// Definitions for utility functions and callbacks -// specific to the DirectSound implementation. - -extern "C" 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; -} - -#include "tchar.h" - -std::string convertTChar( LPCTSTR name ) -{ -#if defined( UNICODE ) || defined( _UNICODE ) - int length = WideCharToMultiByte(CP_UTF8, 0, name, -1, NULL, 0, NULL, NULL); - std::string s( length, 0 ); - length = WideCharToMultiByte(CP_UTF8, 0, name, wcslen(name), &s[0], length, NULL, NULL); -#else - std::string s( name ); -#endif - - return s; -} - -static BOOL CALLBACK deviceQueryCallback( LPGUID lpguid, - LPCTSTR description, - LPCTSTR module, - LPVOID lpContext ) -{ - bool *isInput = (bool *) lpContext; - - HRESULT hr; - bool validDevice = false; - if ( *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 = convertTChar( description ); - if ( name == "Primary Sound Driver" || name == "Primary Sound Capture Driver" ) - name = "Default Device"; - if ( validDevice ) { - for ( unsigned int i=0; i<dsDevices.size(); i++ ) { - if ( dsDevices[i].name == name ) { - dsDevices[i].found = true; - if ( *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 ( *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; } -}; - -extern "C" 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( RtError::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( RtError::WARNING ); - break; - } - if ( subdevice < 0 ) - break; - nDevices++; - } - nextcard: - snd_ctl_close( handle ); - snd_card_next( &card ); - } - - 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; - 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( RtError::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( RtError::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 ); - } - - if ( nDevices == 0 ) { - errorText_ = "RtApiAlsa::getDeviceInfo: no devices found!"; - error( RtError::INVALID_USE ); - } - - if ( device >= nDevices ) { - errorText_ = "RtApiAlsa::getDeviceInfo: device ID is invalid!"; - error( RtError::INVALID_USE ); - } - - 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( RtError::WARNING ); - return info; - } - return devices_[ device ]; - } - - int openMode = SND_PCM_ASYNC; - snd_pcm_stream_t stream; - snd_pcm_info_t *pcminfo; - snd_pcm_info_alloca( &pcminfo ); - snd_pcm_t *phandle; - snd_pcm_hw_params_t *params; - snd_pcm_hw_params_alloca( ¶ms ); - - // First try for playback - stream = SND_PCM_STREAM_PLAYBACK; - snd_pcm_info_set_device( pcminfo, subdevice ); - snd_pcm_info_set_subdevice( pcminfo, 0 ); - snd_pcm_info_set_stream( pcminfo, stream ); - - 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( RtError::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( RtError::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( RtError::WARNING ); - goto captureProbe; - } - info.outputChannels = value; - snd_pcm_close( phandle ); - - captureProbe: - // Now try for capture - stream = SND_PCM_STREAM_CAPTURE; - snd_pcm_info_set_stream( pcminfo, stream ); - - 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; - } - - 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( RtError::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( RtError::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( RtError::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( RtError::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( RtError::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.sampleRates.size() == 0 ) { - snd_pcm_close( phandle ); - errorStream_ << "RtApiAlsa::getDeviceInfo: no supported sample rates found for device (" << name << ")."; - errorText_ = errorStream_.str(); - error( RtError::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 ) { - errorStream_ << "RtApiAlsa::getDeviceInfo: pcm device (" << name << ") data format not supported by RtAudio."; - errorText_ = errorStream_.str(); - error( RtError::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 ); - 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 ); - } - - 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( RtError::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 ) { - struct sched_param param; - int priority = options->priority; - int min = sched_get_priority_min( SCHED_RR ); - int max = sched_get_priority_max( SCHED_RR ); - if ( priority < min ) priority = min; - else if ( priority > max ) priority = max; - param.sched_priority = priority; - pthread_attr_setschedparam( &attr, ¶m ); - pthread_attr_setschedpolicy( &attr, SCHED_RR ); - } - else - pthread_attr_setschedpolicy( &attr, SCHED_OTHER ); -#else - pthread_attr_setschedpolicy( &attr, SCHED_OTHER ); -#endif - - stream_.callbackInfo.isRunning = true; - result = pthread_create( &stream_.callbackInfo.thread, &attr, 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; - } - - return FAILURE; -} - -void RtApiAlsa :: closeStream() -{ - if ( stream_.state == STREAM_CLOSED ) { - errorText_ = "RtApiAlsa::closeStream(): no open stream to close!"; - error( RtError::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( RtError::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 ) { - 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( RtError::SYSTEM_ERROR ); -} - -void RtApiAlsa :: stopStream() -{ - verifyStream(); - if ( stream_.state == STREAM_STOPPED ) { - errorText_ = "RtApiAlsa::stopStream(): the stream is already stopped!"; - error( RtError::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: - MUTEX_UNLOCK( &stream_.mutex ); - - if ( result >= 0 ) return; - error( RtError::SYSTEM_ERROR ); -} - -void RtApiAlsa :: abortStream() -{ - verifyStream(); - if ( stream_.state == STREAM_STOPPED ) { - errorText_ = "RtApiAlsa::abortStream(): the stream is already stopped!"; - error( RtError::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: - MUTEX_UNLOCK( &stream_.mutex ); - - if ( result >= 0 ) return; - error( RtError::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( RtError::WARNING ); - return; - } - - int doStopStream = 0; - RtAudioCallback callback = (RtAudioCallback) stream_.callbackInfo.callback; - double streamTime = getStreamTime(); - RtAudioStreamStatus status = 0; - if ( stream_.mode != INPUT && apiInfo->xrun[0] == true ) { - status |= RTAUDIO_OUTPUT_UNDERFLOW; - apiInfo->xrun[0] = false; - } - if ( stream_.mode != OUTPUT && apiInfo->xrun[1] == true ) { - status |= RTAUDIO_INPUT_OVERFLOW; - apiInfo->xrun[1] = false; - } - doStopStream = callback( stream_.userBuffer[0], stream_.userBuffer[1], - stream_.bufferSize, streamTime, status, stream_.callbackInfo.userData ); - - if ( doStopStream == 2 ) { - abortStream(); - return; - } - - MUTEX_LOCK( &stream_.mutex ); - - // The state might change while waiting on a mutex. - if ( stream_.state == STREAM_STOPPED ) goto unlock; - - int result; - char *buffer; - int channels; - snd_pcm_t **handle; - snd_pcm_sframes_t frames; - RtAudioFormat format; - handle = (snd_pcm_t **) apiInfo->handles; - - if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) { - - // Setup parameters. - if ( stream_.doConvertBuffer[1] ) { - buffer = stream_.deviceBuffer; - channels = stream_.nDeviceChannels[1]; - format = stream_.deviceFormat[1]; - } - else { - buffer = stream_.userBuffer[1]; - channels = stream_.nUserChannels[1]; - format = stream_.userFormat; - } - - // Read samples from device in interleaved/non-interleaved format. - if ( stream_.deviceInterleaved[1] ) - result = snd_pcm_readi( handle[1], buffer, stream_.bufferSize ); - else { - void *bufs[channels]; - size_t offset = stream_.bufferSize * formatBytes( format ); - for ( int i=0; i<channels; i++ ) - bufs[i] = (void *) (buffer + (i * offset)); - result = snd_pcm_readn( handle[1], bufs, stream_.bufferSize ); - } - - if ( result < (int) stream_.bufferSize ) { - // Either an error or overrun occured. - if ( result == -EPIPE ) { - snd_pcm_state_t state = snd_pcm_state( handle[1] ); - if ( state == SND_PCM_STATE_XRUN ) { - apiInfo->xrun[1] = true; - result = snd_pcm_prepare( handle[1] ); - if ( result < 0 ) { - errorStream_ << "RtApiAlsa::callbackEvent: error preparing device after overrun, " << snd_strerror( result ) << "."; - errorText_ = errorStream_.str(); - } - } - else { - errorStream_ << "RtApiAlsa::callbackEvent: error, current state is " << snd_pcm_state_name( state ) << ", " << snd_strerror( result ) << "."; - errorText_ = errorStream_.str(); - } - } - else { - errorStream_ << "RtApiAlsa::callbackEvent: audio read error, " << snd_strerror( result ) << "."; - errorText_ = errorStream_.str(); - } - error( RtError::WARNING ); - goto tryOutput; - } - - // Do byte swapping if necessary. - if ( stream_.doByteSwap[1] ) - byteSwapBuffer( buffer, stream_.bufferSize * channels, format ); - - // Do buffer conversion if necessary. - if ( stream_.doConvertBuffer[1] ) - convertBuffer( stream_.userBuffer[1], stream_.deviceBuffer, stream_.convertInfo[1] ); - - // Check stream latency - result = snd_pcm_delay( handle[1], &frames ); - if ( result == 0 && frames > 0 ) stream_.latency[1] = frames; - } - - tryOutput: - - if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { - - // Setup parameters and do buffer conversion if necessary. - if ( stream_.doConvertBuffer[0] ) { - buffer = stream_.deviceBuffer; - convertBuffer( buffer, stream_.userBuffer[0], stream_.convertInfo[0] ); - channels = stream_.nDeviceChannels[0]; - format = stream_.deviceFormat[0]; - } - else { - buffer = stream_.userBuffer[0]; - channels = stream_.nUserChannels[0]; - format = stream_.userFormat; - } - - // Do byte swapping if necessary. - if ( stream_.doByteSwap[0] ) - byteSwapBuffer(buffer, stream_.bufferSize * channels, format); - - // Write samples to device in interleaved/non-interleaved format. - if ( stream_.deviceInterleaved[0] ) - result = snd_pcm_writei( handle[0], buffer, stream_.bufferSize ); - else { - void *bufs[channels]; - size_t offset = stream_.bufferSize * formatBytes( format ); - for ( int i=0; i<channels; i++ ) - bufs[i] = (void *) (buffer + (i * offset)); - result = snd_pcm_writen( handle[0], bufs, stream_.bufferSize ); - } - - if ( result < (int) stream_.bufferSize ) { - // Either an error or underrun occured. - if ( result == -EPIPE ) { - snd_pcm_state_t state = snd_pcm_state( handle[0] ); - if ( state == SND_PCM_STATE_XRUN ) { - apiInfo->xrun[0] = true; - result = snd_pcm_prepare( handle[0] ); - if ( result < 0 ) { - errorStream_ << "RtApiAlsa::callbackEvent: error preparing device after underrun, " << snd_strerror( result ) << "."; - errorText_ = errorStream_.str(); - } - } - else { - 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( RtError::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(); -} - -extern "C" void *alsaCallbackHandler( void *ptr ) -{ - CallbackInfo *info = (CallbackInfo *) ptr; - RtApiAlsa *object = (RtApiAlsa *) info->object; - bool *isRunning = &info->isRunning; - - 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> - -namespace { -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.nativeFormats = RTAUDIO_SINT16 | RTAUDIO_SINT32 | RTAUDIO_FLOAT32; - - return info; -} - -extern "C" 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( RtError::WARNING ); - return; - } - - RtAudioCallback callback = (RtAudioCallback) stream_.callbackInfo.callback; - double streamTime = getStreamTime(); - RtAudioStreamStatus status = 0; - int doStopStream = callback( stream_.userBuffer[0], stream_.userBuffer[1], - stream_.bufferSize, streamTime, status, - stream_.callbackInfo.userData ); - - if ( doStopStream == 2 ) { - abortStream(); - return; - } - - MUTEX_LOCK( &stream_.mutex ); - - if ( stream_.state != STREAM_RUNNING ) - goto unlock; - - int pa_error; - size_t bytes; - switch ( stream_.mode ) { - case INPUT: - bytes = stream_.nUserChannels[1] * stream_.bufferSize * formatBytes( stream_.userFormat ); - if ( pa_simple_read( pah->s_rec, stream_.userBuffer[1], bytes, &pa_error ) < 0 ) { - errorStream_ << "RtApiPulse::callbackEvent: audio read error, " << - pa_strerror( pa_error ) << "."; - errorText_ = errorStream_.str(); - error( RtError::WARNING ); - } - break; - case OUTPUT: - bytes = stream_.nUserChannels[0] * stream_.bufferSize * formatBytes( stream_.userFormat ); - if ( pa_simple_write( pah->s_play, stream_.userBuffer[0], bytes, &pa_error ) < 0 ) { - errorStream_ << "RtApiPulse::callbackEvent: audio write error, " << - pa_strerror( pa_error ) << "."; - errorText_ = errorStream_.str(); - error( RtError::WARNING ); - } - break; - case DUPLEX: - bytes = stream_.nUserChannels[1] * stream_.bufferSize * formatBytes( stream_.userFormat ); - if ( pa_simple_read( pah->s_rec, stream_.userBuffer[1], bytes, &pa_error ) < 0 ) { - errorStream_ << "RtApiPulse::callbackEvent: audio read error, " << - pa_strerror( pa_error ) << "."; - errorText_ = errorStream_.str(); - error( RtError::WARNING ); - } - bytes = stream_.nUserChannels[0] * stream_.bufferSize * formatBytes( stream_.userFormat ); - if ( pa_simple_write( pah->s_play, stream_.userBuffer[0], bytes, &pa_error ) < 0) { - errorStream_ << "RtApiPulse::callbackEvent: audio write error, " << - pa_strerror( pa_error ) << "."; - errorText_ = errorStream_.str(); - error( RtError::WARNING ); - } - break; - default: - // ERROR - break; - } - - 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( RtError::INVALID_USE ); - return; - } - if ( stream_.state == STREAM_RUNNING ) { - errorText_ = "RtApiPulse::startStream(): the stream is already running!"; - error( RtError::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( RtError::INVALID_USE ); - return; - } - if ( stream_.state == STREAM_STOPPED ) { - errorText_ = "RtApiPulse::stopStream(): the stream is already stopped!"; - error( RtError::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( RtError::SYSTEM_ERROR ); - } - } - - 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( RtError::INVALID_USE ); - return; - } - if ( stream_.state == STREAM_STOPPED ) { - errorText_ = "RtApiPulse::abortStream(): the stream is already stopped!"; - error( RtError::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( RtError::SYSTEM_ERROR ); - } - } - - 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; - ss.format = sf->pa_format; - break; - } - } - if ( !sf_found ) { - errorText_ = "RtApiPulse::probeDeviceOpen: unsupported sample format."; - return false; - } - - if ( options && ( options->flags & RTAUDIO_NONINTERLEAVED ) ) { - errorText_ = "RtApiPulse::probeDeviceOpen: only interleaved audio data supported."; - return false; - } - - stream_.userInterleaved = true; - stream_.nBuffers = 1; - - stream_.deviceInterleaved[mode] = true; - stream_.doByteSwap[mode] = false; - stream_.doConvertBuffer[mode] = false; - stream_.deviceFormat[mode] = stream_.userFormat; - stream_.nUserChannels[mode] = channels; - stream_.nDeviceChannels[mode] = channels; - stream_.channelOffset[mode] = 0; - - // 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_.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; - switch ( mode ) { - case INPUT: - pah->s_rec = pa_simple_new( NULL, "RtAudio", PA_STREAM_RECORD, NULL, "Record", &ss, NULL, NULL, &error ); - if ( !pah->s_rec ) { - errorText_ = "RtApiPulse::probeDeviceOpen: error connecting input to PulseAudio server."; - goto error; - } - break; - case OUTPUT: - pah->s_play = pa_simple_new( NULL, "RtAudio", PA_STREAM_PLAYBACK, NULL, "Playback", &ss, NULL, NULL, &error ); - if ( !pah->s_play ) { - errorText_ = "RtApiPulse::probeDeviceOpen: error connecting output to PulseAudio server."; - goto error; - } - break; - default: - goto error; - } - - if ( stream_.mode == UNINITIALIZED ) - stream_.mode = mode; - else if ( stream_.mode == mode ) - goto error; - else - stream_.mode = DUPLEX; - - stream_.state = STREAM_STOPPED; - - 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; - } - } - return true; - - error: - closeStream(); - return false; -} - -//******************** End of __LINUX_PULSE__ *********************// -#endif - -#if defined(__LINUX_OSS__) - -#include <unistd.h> -#include <sys/ioctl.h> -#include <unistd.h> -#include <fcntl.h> -#include "soundcard.h" -#include <errno.h> -#include <math.h> - -extern "C" 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( RtError::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( RtError::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( RtError::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( RtError::WARNING ); - return info; - } - - unsigned nDevices = sysinfo.numaudios; - if ( nDevices == 0 ) { - close( mixerfd ); - errorText_ = "RtApiOss::getDeviceInfo: no devices found!"; - error( RtError::INVALID_USE ); - } - - if ( device >= nDevices ) { - close( mixerfd ); - errorText_ = "RtApiOss::getDeviceInfo: device ID is invalid!"; - error( RtError::INVALID_USE ); - } - - 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( RtError::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( RtError::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] ); - 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.sampleRates.size() == 0 ) { - errorStream_ << "RtApiOss::getDeviceInfo: no supported sample rates found for device (" << ainfo.name << ")."; - errorText_ = errorStream_.str(); - error( RtError::WARNING ); - } - else { - info.probed = true; - info.name = ainfo.name; - } - - return info; -} - - -bool RtApiOss :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels, - unsigned int firstChannel, unsigned int sampleRate, - RtAudioFormat format, unsigned int *bufferSize, - RtAudio::StreamOptions *options ) -{ - int mixerfd = open( "/dev/mixer", O_RDWR, 0 ); - if ( mixerfd == -1 ) { - errorText_ = "RtApiOss::probeDeviceOpen: error opening '/dev/mixer'."; - return FAILURE; - } - - oss_sysinfo sysinfo; - int result = ioctl( mixerfd, SNDCTL_SYSINFO, &sysinfo ); - if ( result == -1 ) { - close( mixerfd ); - errorText_ = "RtApiOss::probeDeviceOpen: error getting sysinfo, OSS version >= 4.0 is required."; - return FAILURE; - } - - unsigned nDevices = sysinfo.numaudios; - if ( nDevices == 0 ) { - // This should not happen because a check is made before this function is called. - close( mixerfd ); - errorText_ = "RtApiOss::probeDeviceOpen: no devices found!"; - return FAILURE; - } - - if ( device >= nDevices ) { - // This should not happen because a check is made before this function is called. - close( mixerfd ); - errorText_ = "RtApiOss::probeDeviceOpen: device ID is invalid!"; - return FAILURE; - } - - oss_audioinfo ainfo; - ainfo.dev = device; - result = ioctl( mixerfd, SNDCTL_AUDIOINFO, &ainfo ); - close( mixerfd ); - if ( result == -1 ) { - errorStream_ << "RtApiOss::getDeviceInfo: error getting device (" << ainfo.name << ") info."; - errorText_ = errorStream_.str(); - return FAILURE; - } - - // Check if device supports input or output - if ( ( mode == OUTPUT && !( ainfo.caps & PCM_CAP_OUTPUT ) ) || - ( mode == INPUT && !( ainfo.caps & PCM_CAP_INPUT ) ) ) { - if ( mode == OUTPUT ) - errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") does not support output."; - else - errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") does not support input."; - errorText_ = errorStream_.str(); - return FAILURE; - } - - int flags = 0; - OssHandle *handle = (OssHandle *) stream_.apiHandle; - if ( mode == OUTPUT ) - flags |= O_WRONLY; - else { // mode == INPUT - if (stream_.mode == OUTPUT && stream_.device[0] == device) { - // We just set the same device for playback ... close and reopen for duplex (OSS only). - close( handle->id[0] ); - handle->id[0] = 0; - if ( !( ainfo.caps & PCM_CAP_DUPLEX ) ) { - errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") does not support duplex mode."; - errorText_ = errorStream_.str(); - return FAILURE; - } - // Check that the number previously set channels is the same. - if ( stream_.nUserChannels[0] != channels ) { - errorStream_ << "RtApiOss::probeDeviceOpen: input/output channels must be equal for OSS duplex device (" << ainfo.name << ")."; - errorText_ = errorStream_.str(); - return FAILURE; - } - flags |= O_RDWR; - } - else - flags |= O_RDONLY; - } - - // Set exclusive access if specified. - if ( options && options->flags & RTAUDIO_HOG_DEVICE ) flags |= O_EXCL; - - // Try to open the device. - int fd; - fd = open( ainfo.devnode, flags, 0 ); - if ( fd == -1 ) { - if ( errno == EBUSY ) - errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") is busy."; - else - errorStream_ << "RtApiOss::probeDeviceOpen: error opening device (" << ainfo.name << ")."; - errorText_ = errorStream_.str(); - return FAILURE; - } - - // For duplex operation, specifically set this mode (this doesn't seem to work). - /* - if ( flags | O_RDWR ) { - result = ioctl( fd, SNDCTL_DSP_SETDUPLEX, NULL ); - if ( result == -1) { - errorStream_ << "RtApiOss::probeDeviceOpen: error setting duplex mode for device (" << ainfo.name << ")."; - errorText_ = errorStream_.str(); - return FAILURE; - } - } - */ - - // Check the device channel support. - stream_.nUserChannels[mode] = channels; - if ( ainfo.max_channels < (int)(channels + firstChannel) ) { - close( fd ); - errorStream_ << "RtApiOss::probeDeviceOpen: the device (" << ainfo.name << ") does not support requested channel parameters."; - errorText_ = errorStream_.str(); - return FAILURE; - } - - // Set the number of channels. - int deviceChannels = channels + firstChannel; - result = ioctl( fd, SNDCTL_DSP_CHANNELS, &deviceChannels ); - if ( result == -1 || deviceChannels < (int)(channels + firstChannel) ) { - close( fd ); - errorStream_ << "RtApiOss::probeDeviceOpen: error setting channel parameters on device (" << ainfo.name << ")."; - errorText_ = errorStream_.str(); - return FAILURE; - } - stream_.nDeviceChannels[mode] = deviceChannels; - - // Get the data format mask - int mask; - result = ioctl( fd, SNDCTL_DSP_GETFMTS, &mask ); - if ( result == -1 ) { - close( fd ); - errorStream_ << "RtApiOss::probeDeviceOpen: error getting device (" << ainfo.name << ") data formats."; - errorText_ = errorStream_.str(); - return FAILURE; - } - - // Determine how to set the device format. - stream_.userFormat = format; - int deviceFormat = -1; - stream_.doByteSwap[mode] = false; - if ( format == RTAUDIO_SINT8 ) { - if ( mask & AFMT_S8 ) { - deviceFormat = AFMT_S8; - stream_.deviceFormat[mode] = RTAUDIO_SINT8; - } - } - else if ( format == RTAUDIO_SINT16 ) { - if ( mask & AFMT_S16_NE ) { - deviceFormat = AFMT_S16_NE; - stream_.deviceFormat[mode] = RTAUDIO_SINT16; - } - else if ( mask & AFMT_S16_OE ) { - deviceFormat = AFMT_S16_OE; - stream_.deviceFormat[mode] = RTAUDIO_SINT16; - stream_.doByteSwap[mode] = true; - } - } - else if ( format == RTAUDIO_SINT24 ) { - if ( mask & AFMT_S24_NE ) { - deviceFormat = AFMT_S24_NE; - stream_.deviceFormat[mode] = RTAUDIO_SINT24; - } - else if ( mask & AFMT_S24_OE ) { - deviceFormat = AFMT_S24_OE; - stream_.deviceFormat[mode] = RTAUDIO_SINT24; - stream_.doByteSwap[mode] = true; - } - } - else if ( format == RTAUDIO_SINT32 ) { - if ( mask & AFMT_S32_NE ) { - deviceFormat = AFMT_S32_NE; - stream_.deviceFormat[mode] = RTAUDIO_SINT32; - } - else if ( mask & AFMT_S32_OE ) { - deviceFormat = AFMT_S32_OE; - stream_.deviceFormat[mode] = RTAUDIO_SINT32; - stream_.doByteSwap[mode] = true; - } - } - - if ( deviceFormat == -1 ) { - // The user requested format is not natively supported by the device. - if ( mask & AFMT_S16_NE ) { - deviceFormat = AFMT_S16_NE; - stream_.deviceFormat[mode] = RTAUDIO_SINT16; - } - else if ( mask & AFMT_S32_NE ) { - deviceFormat = AFMT_S32_NE; - stream_.deviceFormat[mode] = RTAUDIO_SINT32; - } - else if ( mask & AFMT_S24_NE ) { - deviceFormat = AFMT_S24_NE; - stream_.deviceFormat[mode] = RTAUDIO_SINT24; - } - else if ( mask & AFMT_S16_OE ) { - deviceFormat = AFMT_S16_OE; - stream_.deviceFormat[mode] = RTAUDIO_SINT16; - stream_.doByteSwap[mode] = true; - } - else if ( mask & AFMT_S32_OE ) { - deviceFormat = AFMT_S32_OE; - stream_.deviceFormat[mode] = RTAUDIO_SINT32; - stream_.doByteSwap[mode] = true; - } - else if ( mask & AFMT_S24_OE ) { - deviceFormat = AFMT_S24_OE; - stream_.deviceFormat[mode] = RTAUDIO_SINT24; - stream_.doByteSwap[mode] = true; - } - else if ( mask & AFMT_S8) { - deviceFormat = AFMT_S8; - stream_.deviceFormat[mode] = RTAUDIO_SINT8; - } - } - - if ( stream_.deviceFormat[mode] == 0 ) { - // This really shouldn't happen ... - close( fd ); - errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") data format not supported by RtAudio."; - errorText_ = errorStream_.str(); - return FAILURE; - } - - // Set the data format. - int temp = deviceFormat; - result = ioctl( fd, SNDCTL_DSP_SETFMT, &deviceFormat ); - if ( result == -1 || deviceFormat != temp ) { - close( fd ); - errorStream_ << "RtApiOss::probeDeviceOpen: error setting data format on device (" << ainfo.name << ")."; - errorText_ = errorStream_.str(); - return FAILURE; - } - - // Attempt to set the buffer size. According to OSS, the minimum - // number of buffers is two. The supposed minimum buffer size is 16 - // bytes, so that will be our lower bound. The argument to this - // call is in the form 0xMMMMSSSS (hex), where the buffer size (in - // bytes) is given as 2^SSSS and the number of buffers as 2^MMMM. - // We'll check the actual value used near the end of the setup - // procedure. - int ossBufferBytes = *bufferSize * formatBytes( stream_.deviceFormat[mode] ) * deviceChannels; - if ( ossBufferBytes < 16 ) ossBufferBytes = 16; - int buffers = 0; - if ( options ) buffers = options->numberOfBuffers; - if ( options && options->flags & RTAUDIO_MINIMIZE_LATENCY ) buffers = 2; - if ( buffers < 2 ) buffers = 3; - temp = ((int) buffers << 16) + (int)( log10( (double)ossBufferBytes ) / log10( 2.0 ) ); - result = ioctl( fd, SNDCTL_DSP_SETFRAGMENT, &temp ); - if ( result == -1 ) { - close( fd ); - errorStream_ << "RtApiOss::probeDeviceOpen: error setting buffer size on device (" << ainfo.name << ")."; - errorText_ = errorStream_.str(); - return FAILURE; - } - stream_.nBuffers = buffers; - - // Save buffer size (in sample frames). - *bufferSize = ossBufferBytes / ( formatBytes(stream_.deviceFormat[mode]) * deviceChannels ); - stream_.bufferSize = *bufferSize; - - // Set the sample rate. - int srate = sampleRate; - result = ioctl( fd, SNDCTL_DSP_SPEED, &srate ); - if ( result == -1 ) { - close( fd ); - errorStream_ << "RtApiOss::probeDeviceOpen: error setting sample rate (" << sampleRate << ") on device (" << ainfo.name << ")."; - errorText_ = errorStream_.str(); - return FAILURE; - } - - // Verify the sample rate setup worked. - if ( abs( srate - sampleRate ) > 100 ) { - close( fd ); - errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") does not support sample rate (" << sampleRate << ")."; - errorText_ = errorStream_.str(); - return FAILURE; - } - stream_.sampleRate = sampleRate; - - if ( mode == INPUT && stream_.mode == OUTPUT && stream_.device[0] == device) { - // We're doing duplex setup here. - stream_.deviceFormat[0] = stream_.deviceFormat[1]; - stream_.nDeviceChannels[0] = deviceChannels; - } - - // Set interleaving parameters. - stream_.userInterleaved = true; - stream_.deviceInterleaved[mode] = true; - if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) - stream_.userInterleaved = false; - - // Set flags for buffer conversion - stream_.doConvertBuffer[mode] = false; - if ( stream_.userFormat != stream_.deviceFormat[mode] ) - stream_.doConvertBuffer[mode] = true; - if ( stream_.nUserChannels[mode] < stream_.nDeviceChannels[mode] ) - stream_.doConvertBuffer[mode] = true; - if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] && - stream_.nUserChannels[mode] > 1 ) - stream_.doConvertBuffer[mode] = true; - - // Allocate the stream handles if necessary and then save. - if ( stream_.apiHandle == 0 ) { - try { - handle = new OssHandle; - } - catch ( std::bad_alloc& ) { - errorText_ = "RtApiOss::probeDeviceOpen: error allocating OssHandle memory."; - goto error; - } - - if ( pthread_cond_init( &handle->runnable, NULL ) ) { - errorText_ = "RtApiOss::probeDeviceOpen: error initializing pthread condition variable."; - goto error; - } - - stream_.apiHandle = (void *) handle; - } - else { - handle = (OssHandle *) stream_.apiHandle; - } - handle->id[mode] = fd; - - // Allocate necessary internal buffers. - unsigned long bufferBytes; - bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat ); - stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 ); - if ( stream_.userBuffer[mode] == NULL ) { - errorText_ = "RtApiOss::probeDeviceOpen: error allocating user buffer memory."; - goto error; - } - - if ( stream_.doConvertBuffer[mode] ) { - - bool makeBuffer = true; - bufferBytes = stream_.nDeviceChannels[mode] * formatBytes( stream_.deviceFormat[mode] ); - if ( mode == INPUT ) { - if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) { - unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] ); - if ( bufferBytes <= bytesOut ) makeBuffer = false; - } - } - - if ( makeBuffer ) { - bufferBytes *= *bufferSize; - if ( stream_.deviceBuffer ) free( stream_.deviceBuffer ); - stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 ); - if ( stream_.deviceBuffer == NULL ) { - errorText_ = "RtApiOss::probeDeviceOpen: error allocating device buffer memory."; - goto error; - } - } - } - - stream_.device[mode] = device; - stream_.state = STREAM_STOPPED; - - // Setup the buffer conversion information structure. - if ( stream_.doConvertBuffer[mode] ) setConvertInfo( mode, firstChannel ); - - // Setup thread if necessary. - if ( stream_.mode == OUTPUT && mode == INPUT ) { - // We had already set up an output stream. - stream_.mode = DUPLEX; - if ( stream_.device[0] == device ) handle->id[0] = fd; - } - else { - stream_.mode = mode; - - // Setup callback thread. - stream_.callbackInfo.object = (void *) this; - - // Set the thread attributes for joinable and realtime scheduling - // priority. The higher priority will only take affect if the - // program is run as root or suid. - pthread_attr_t attr; - pthread_attr_init( &attr ); - pthread_attr_setdetachstate( &attr, PTHREAD_CREATE_JOINABLE ); -#ifdef SCHED_RR // Undefined with some OSes (eg: NetBSD 1.6.x with GNU Pthread) - if ( options && options->flags & RTAUDIO_SCHEDULE_REALTIME ) { - struct sched_param param; - int priority = options->priority; - int min = sched_get_priority_min( SCHED_RR ); - int max = sched_get_priority_max( SCHED_RR ); - if ( priority < min ) priority = min; - else if ( priority > max ) priority = max; - param.sched_priority = priority; - pthread_attr_setschedparam( &attr, ¶m ); - pthread_attr_setschedpolicy( &attr, SCHED_RR ); - } - else - pthread_attr_setschedpolicy( &attr, SCHED_OTHER ); -#else - pthread_attr_setschedpolicy( &attr, SCHED_OTHER ); -#endif - - stream_.callbackInfo.isRunning = true; - result = pthread_create( &stream_.callbackInfo.thread, &attr, ossCallbackHandler, &stream_.callbackInfo ); - pthread_attr_destroy( &attr ); - if ( result ) { - stream_.callbackInfo.isRunning = false; - errorText_ = "RtApiOss::error creating callback thread!"; - goto error; - } - } - - return SUCCESS; - - error: - if ( handle ) { - pthread_cond_destroy( &handle->runnable ); - if ( handle->id[0] ) close( handle->id[0] ); - if ( handle->id[1] ) close( handle->id[1] ); - delete handle; - stream_.apiHandle = 0; - } - - for ( int i=0; i<2; i++ ) { - if ( stream_.userBuffer[i] ) { - free( stream_.userBuffer[i] ); - stream_.userBuffer[i] = 0; - } - } - - if ( stream_.deviceBuffer ) { - free( stream_.deviceBuffer ); - stream_.deviceBuffer = 0; - } - - return FAILURE; -} - -void RtApiOss :: closeStream() -{ - if ( stream_.state == STREAM_CLOSED ) { - errorText_ = "RtApiOss::closeStream(): no open stream to close!"; - error( RtError::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( RtError::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( RtError::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( RtError::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( RtError::SYSTEM_ERROR ); -} - -void RtApiOss :: abortStream() -{ - verifyStream(); - if ( stream_.state == STREAM_STOPPED ) { - errorText_ = "RtApiOss::abortStream(): the stream is already stopped!"; - error( RtError::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( RtError::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( RtError::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( RtError::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( RtError::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(); -} - -extern "C" 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( RtError::Type type ) -{ - errorStream_.str(""); // clear the ostringstream - if ( type == RtError::WARNING && showWarnings_ == true ) - std::cerr << '\n' << errorText_ << "\n\n"; - else if ( type != RtError::WARNING ) - throw( RtError( errorText_, type ) ); -} - -void RtApi :: verifyStream() -{ - if ( stream_.state == STREAM_CLOSED ) { - errorText_ = "RtApi:: a stream is not open!"; - error( RtError::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; - 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_SINT24 || format == RTAUDIO_SINT32 || - format == RTAUDIO_FLOAT32 ) - return 4; - else if ( format == RTAUDIO_FLOAT64 ) - return 8; - else if ( format == RTAUDIO_SINT8 ) - return 1; - - errorText_ = "RtApi::formatBytes: undefined format."; - error( RtError::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) { - Int32 *in = (Int32 *)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]] & 0x00ffffff); - 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) { - Int32 *in = (Int32 *)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]] & 0x00ffffff); - 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) { // Hmmm ... we could just leave it in the lower 3 bytes - 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]]; - 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) { - 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]] <<= 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]]; - out[info.outOffset[j]] <<= 8; - } - in += info.inJump; - out += info.outJump; - } - } - else if (info.inFormat == RTAUDIO_SINT24) { - // 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_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]]; - 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) { - 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]] >> 8) & 0x0000ffff); - } - 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) { - 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]] >> 16) & 0x000000ff); - } - in += info.inJump; - out += info.outJump; - } - } - else if (info.inFormat == RTAUDIO_SINT32) { - Int32 *in = (Int32 *)inBuffer; - for (unsigned int i=0; i<stream_.bufferSize; i++) { - for (j=0; j<info.channels; j++) { - out[info.outOffset[j]] = (signed char) ((in[info.inOffset[j]] >> 24) & 0x000000ff); - } - in += info.inJump; - out += info.outJump; - } - } - else if (info.inFormat == RTAUDIO_FLOAT32) { - Float32 *in = (Float32 *)inBuffer; - for (unsigned int i=0; i<stream_.bufferSize; i++) { - for (j=0; j<info.channels; j++) { - out[info.outOffset[j]] = (signed char) (in[info.inOffset[j]] * 127.5 - 0.5); - } - in += info.inJump; - out += info.outJump; - } - } - else if (info.inFormat == RTAUDIO_FLOAT64) { - Float64 *in = (Float64 *)inBuffer; - for (unsigned int i=0; i<stream_.bufferSize; i++) { - for (j=0; j<info.channels; j++) { - out[info.outOffset[j]] = (signed char) (in[info.inOffset[j]] * 127.5 - 0.5); - } - in += info.inJump; - out += info.outJump; - } - } - } -} - - //static inline uint16_t bswap_16(uint16_t x) { return (x>>8) | (x<<8); } - //static inline uint32_t bswap_32(uint32_t x) { return (bswap_16(x&0xffff)<<16) | (bswap_16(x>>16)); } - //static inline uint64_t bswap_64(uint64_t x) { return (((unsigned long long)bswap_32(x&0xffffffffull))<<32) | (bswap_32(x>>32)); } - -void RtApi :: byteSwapBuffer( char *buffer, unsigned int samples, RtAudioFormat format ) -{ - register char val; - register char *ptr; - - ptr = buffer; - if ( format == RTAUDIO_SINT16 ) { - for ( unsigned int i=0; i<samples; i++ ) { - // Swap 1st and 2nd bytes. - val = *(ptr); - *(ptr) = *(ptr+1); - *(ptr+1) = val; - - // Increment 2 bytes. - ptr += 2; - } - } - else if ( format == RTAUDIO_SINT24 || - 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; + (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) 2001-2019 Gary P. Scavone + + Permission is hereby granted, free of charge, to any person + obtaining a copy of this software and associated documentation files + (the "Software"), to deal in the Software without restriction, + including without limitation the rights to use, copy, modify, merge, + publish, distribute, sublicense, and/or sell copies of the Software, + and to permit persons to whom the Software is furnished to do so, + subject to the following conditions: + + The above copyright notice and this permission notice shall be + included in all copies or substantial portions of the Software. + + Any person wishing to distribute modifications to the Software is + asked to send the modifications to the original developer so that + they can be incorporated into the canonical version. This is, + however, not a binding provision of this license. + + THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, + EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF + MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. + IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR + ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF + CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION + WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE. +*/ +/************************************************************************/ + +// RtAudio: Version 5.1.0 + +#include "RtAudio.h" +#include <iostream> +#include <cstdlib> +#include <cstring> +#include <climits> +#include <cmath> +#include <algorithm> + +// Static variable definitions. +const unsigned int RtApi::MAX_SAMPLE_RATES = 14; +const unsigned int RtApi::SAMPLE_RATES[] = { + 4000, 5512, 8000, 9600, 11025, 16000, 22050, + 32000, 44100, 48000, 88200, 96000, 176400, 192000 +}; + +#if defined(__WINDOWS_DS__) || defined(__WINDOWS_ASIO__) || defined(__WINDOWS_WASAPI__) + #define MUTEX_INITIALIZE(A) InitializeCriticalSection(A) + #define MUTEX_DESTROY(A) DeleteCriticalSection(A) + #define MUTEX_LOCK(A) EnterCriticalSection(A) + #define MUTEX_UNLOCK(A) LeaveCriticalSection(A) + + #include "tchar.h" + + static std::string convertCharPointerToStdString(const char *text) + { + return std::string(text); + } + + static std::string convertCharPointerToStdString(const wchar_t *text) + { + int length = WideCharToMultiByte(CP_UTF8, 0, text, -1, NULL, 0, NULL, NULL); + std::string s( length-1, '\0' ); + WideCharToMultiByte(CP_UTF8, 0, text, -1, &s[0], length, NULL, NULL); + return s; + } + +#elif defined(__LINUX_ALSA__) || defined(__LINUX_PULSE__) || defined(__UNIX_JACK__) || defined(__LINUX_OSS__) || defined(__MACOSX_CORE__) + // pthread API + #define MUTEX_INITIALIZE(A) pthread_mutex_init(A, NULL) + #define MUTEX_DESTROY(A) pthread_mutex_destroy(A) + #define MUTEX_LOCK(A) pthread_mutex_lock(A) + #define MUTEX_UNLOCK(A) pthread_mutex_unlock(A) +#else + #define MUTEX_INITIALIZE(A) abs(*A) // dummy definitions + #define MUTEX_DESTROY(A) abs(*A) // dummy definitions +#endif + +// *************************************************** // +// +// RtAudio definitions. +// +// *************************************************** // + +std::string RtAudio :: getVersion( void ) +{ + return RTAUDIO_VERSION; +} + +// Define API names and display names. +// Must be in same order as API enum. +extern "C" { +const char* rtaudio_api_names[][2] = { + { "unspecified" , "Unknown" }, + { "alsa" , "ALSA" }, + { "pulse" , "Pulse" }, + { "oss" , "OpenSoundSystem" }, + { "jack" , "Jack" }, + { "core" , "CoreAudio" }, + { "wasapi" , "WASAPI" }, + { "asio" , "ASIO" }, + { "ds" , "DirectSound" }, + { "dummy" , "Dummy" }, +}; +const unsigned int rtaudio_num_api_names = + sizeof(rtaudio_api_names)/sizeof(rtaudio_api_names[0]); + +// The order here will control the order of RtAudio's API search in +// the constructor. +extern "C" const RtAudio::Api rtaudio_compiled_apis[] = { +#if defined(__UNIX_JACK__) + RtAudio::UNIX_JACK, +#endif +#if defined(__LINUX_PULSE__) + RtAudio::LINUX_PULSE, +#endif +#if defined(__LINUX_ALSA__) + RtAudio::LINUX_ALSA, +#endif +#if defined(__LINUX_OSS__) + RtAudio::LINUX_OSS, +#endif +#if defined(__WINDOWS_ASIO__) + RtAudio::WINDOWS_ASIO, +#endif +#if defined(__WINDOWS_WASAPI__) + RtAudio::WINDOWS_WASAPI, +#endif +#if defined(__WINDOWS_DS__) + RtAudio::WINDOWS_DS, +#endif +#if defined(__MACOSX_CORE__) + RtAudio::MACOSX_CORE, +#endif +#if defined(__RTAUDIO_DUMMY__) + RtAudio::RTAUDIO_DUMMY, +#endif + RtAudio::UNSPECIFIED, +}; +extern "C" const unsigned int rtaudio_num_compiled_apis = + sizeof(rtaudio_compiled_apis)/sizeof(rtaudio_compiled_apis[0])-1; +} + +// This is a compile-time check that rtaudio_num_api_names == RtAudio::NUM_APIS. +// If the build breaks here, check that they match. +template<bool b> class StaticAssert { private: StaticAssert() {} }; +template<> class StaticAssert<true>{ public: StaticAssert() {} }; +class StaticAssertions { StaticAssertions() { + StaticAssert<rtaudio_num_api_names == RtAudio::NUM_APIS>(); +}}; + +void RtAudio :: getCompiledApi( std::vector<RtAudio::Api> &apis ) +{ + apis = std::vector<RtAudio::Api>(rtaudio_compiled_apis, + rtaudio_compiled_apis + rtaudio_num_compiled_apis); +} + +std::string RtAudio :: getApiName( RtAudio::Api api ) +{ + if (api < 0 || api >= RtAudio::NUM_APIS) + return ""; + return rtaudio_api_names[api][0]; +} + +std::string RtAudio :: getApiDisplayName( RtAudio::Api api ) +{ + if (api < 0 || api >= RtAudio::NUM_APIS) + return "Unknown"; + return rtaudio_api_names[api][1]; +} + +RtAudio::Api RtAudio :: getCompiledApiByName( const std::string &name ) +{ + unsigned int i=0; + for (i = 0; i < rtaudio_num_compiled_apis; ++i) + if (name == rtaudio_api_names[rtaudio_compiled_apis[i]][0]) + return rtaudio_compiled_apis[i]; + return RtAudio::UNSPECIFIED; +} + +void RtAudio :: openRtApi( RtAudio::Api api ) +{ + if ( rtapi_ ) + delete rtapi_; + rtapi_ = 0; + +#if defined(__UNIX_JACK__) + if ( api == UNIX_JACK ) + rtapi_ = new RtApiJack(); +#endif +#if defined(__LINUX_ALSA__) + if ( api == LINUX_ALSA ) + rtapi_ = new RtApiAlsa(); +#endif +#if defined(__LINUX_PULSE__) + if ( api == LINUX_PULSE ) + rtapi_ = new RtApiPulse(); +#endif +#if defined(__LINUX_OSS__) + if ( api == LINUX_OSS ) + rtapi_ = new RtApiOss(); +#endif +#if defined(__WINDOWS_ASIO__) + if ( api == WINDOWS_ASIO ) + rtapi_ = new RtApiAsio(); +#endif +#if defined(__WINDOWS_WASAPI__) + if ( api == WINDOWS_WASAPI ) + rtapi_ = new RtApiWasapi(); +#endif +#if defined(__WINDOWS_DS__) + if ( api == WINDOWS_DS ) + rtapi_ = new RtApiDs(); +#endif +#if defined(__MACOSX_CORE__) + if ( api == MACOSX_CORE ) + rtapi_ = new RtApiCore(); +#endif +#if defined(__RTAUDIO_DUMMY__) + if ( api == RTAUDIO_DUMMY ) + rtapi_ = new RtApiDummy(); +#endif +} + +RtAudio :: RtAudio( RtAudio::Api api ) +{ + rtapi_ = 0; + + if ( api != UNSPECIFIED ) { + // Attempt to open the specified API. + openRtApi( api ); + if ( rtapi_ ) return; + + // No compiled support for specified API value. Issue a debug + // warning and continue as if no API was specified. + std::cerr << "\nRtAudio: no compiled support for specified API argument!\n" << std::endl; + } + + // Iterate through the compiled APIs and return as soon as we find + // one with at least one device or we reach the end of the list. + std::vector< RtAudio::Api > apis; + getCompiledApi( apis ); + for ( unsigned int i=0; i<apis.size(); i++ ) { + openRtApi( apis[i] ); + if ( rtapi_ && rtapi_->getDeviceCount() ) break; + } + + if ( rtapi_ ) return; + + // It should not be possible to get here because the preprocessor + // definition __RTAUDIO_DUMMY__ is automatically defined if no + // API-specific definitions are passed to the compiler. But just in + // case something weird happens, we'll thow an error. + std::string errorText = "\nRtAudio: no compiled API support found ... critical error!!\n\n"; + throw( RtAudioError( errorText, RtAudioError::UNSPECIFIED ) ); +} + +RtAudio :: ~RtAudio() +{ + if ( rtapi_ ) + delete rtapi_; +} + +void RtAudio :: openStream( RtAudio::StreamParameters *outputParameters, + RtAudio::StreamParameters *inputParameters, + RtAudioFormat format, unsigned int sampleRate, + unsigned int *bufferFrames, + RtAudioCallback callback, void *userData, + RtAudio::StreamOptions *options, + RtAudioErrorCallback errorCallback ) +{ + return rtapi_->openStream( outputParameters, inputParameters, format, + sampleRate, bufferFrames, callback, + userData, options, errorCallback ); +} + +// *************************************************** // +// +// Public RtApi definitions (see end of file for +// private or protected utility functions). +// +// *************************************************** // + +RtApi :: RtApi() +{ + stream_.state = STREAM_CLOSED; + stream_.mode = UNINITIALIZED; + stream_.apiHandle = 0; + stream_.userBuffer[0] = 0; + stream_.userBuffer[1] = 0; + MUTEX_INITIALIZE( &stream_.mutex ); + showWarnings_ = true; + firstErrorOccurred_ = false; +} + +RtApi :: ~RtApi() +{ + MUTEX_DESTROY( &stream_.mutex ); +} + +void RtApi :: openStream( RtAudio::StreamParameters *oParams, + RtAudio::StreamParameters *iParams, + RtAudioFormat format, unsigned int sampleRate, + unsigned int *bufferFrames, + RtAudioCallback callback, void *userData, + RtAudio::StreamOptions *options, + RtAudioErrorCallback errorCallback ) +{ + if ( stream_.state != STREAM_CLOSED ) { + errorText_ = "RtApi::openStream: a stream is already open!"; + error( RtAudioError::INVALID_USE ); + return; + } + + // Clear stream information potentially left from a previously open stream. + clearStreamInfo(); + + if ( oParams && oParams->nChannels < 1 ) { + errorText_ = "RtApi::openStream: a non-NULL output StreamParameters structure cannot have an nChannels value less than one."; + error( RtAudioError::INVALID_USE ); + return; + } + + if ( iParams && iParams->nChannels < 1 ) { + errorText_ = "RtApi::openStream: a non-NULL input StreamParameters structure cannot have an nChannels value less than one."; + error( RtAudioError::INVALID_USE ); + return; + } + + if ( oParams == NULL && iParams == NULL ) { + errorText_ = "RtApi::openStream: input and output StreamParameters structures are both NULL!"; + error( RtAudioError::INVALID_USE ); + return; + } + + if ( formatBytes(format) == 0 ) { + errorText_ = "RtApi::openStream: 'format' parameter value is undefined."; + error( RtAudioError::INVALID_USE ); + return; + } + + unsigned int nDevices = getDeviceCount(); + unsigned int oChannels = 0; + if ( oParams ) { + oChannels = oParams->nChannels; + if ( oParams->deviceId >= nDevices ) { + errorText_ = "RtApi::openStream: output device parameter value is invalid."; + error( RtAudioError::INVALID_USE ); + return; + } + } + + unsigned int iChannels = 0; + if ( iParams ) { + iChannels = iParams->nChannels; + if ( iParams->deviceId >= nDevices ) { + errorText_ = "RtApi::openStream: input device parameter value is invalid."; + error( RtAudioError::INVALID_USE ); + return; + } + } + + bool result; + + if ( oChannels > 0 ) { + + result = probeDeviceOpen( oParams->deviceId, OUTPUT, oChannels, oParams->firstChannel, + sampleRate, format, bufferFrames, options ); + if ( result == false ) { + error( RtAudioError::SYSTEM_ERROR ); + return; + } + } + + if ( iChannels > 0 ) { + + result = probeDeviceOpen( iParams->deviceId, INPUT, iChannels, iParams->firstChannel, + sampleRate, format, bufferFrames, options ); + if ( result == false ) { + if ( oChannels > 0 ) closeStream(); + error( RtAudioError::SYSTEM_ERROR ); + return; + } + } + + stream_.callbackInfo.callback = (void *) callback; + stream_.callbackInfo.userData = userData; + stream_.callbackInfo.errorCallback = (void *) errorCallback; + + if ( options ) options->numberOfBuffers = stream_.nBuffers; + stream_.state = STREAM_STOPPED; +} + +unsigned int RtApi :: getDefaultInputDevice( void ) +{ + // Should be implemented in subclasses if possible. + return 0; +} + +unsigned int RtApi :: getDefaultOutputDevice( void ) +{ + // Should be implemented in subclasses if possible. + return 0; +} + +void RtApi :: closeStream( void ) +{ + // MUST be implemented in subclasses! + return; +} + +bool RtApi :: probeDeviceOpen( unsigned int /*device*/, StreamMode /*mode*/, unsigned int /*channels*/, + unsigned int /*firstChannel*/, unsigned int /*sampleRate*/, + RtAudioFormat /*format*/, unsigned int * /*bufferSize*/, + RtAudio::StreamOptions * /*options*/ ) +{ + // MUST be implemented in subclasses! + return FAILURE; +} + +void RtApi :: tickStreamTime( void ) +{ + // Subclasses that do not provide their own implementation of + // getStreamTime should call this function once per buffer I/O to + // provide basic stream time support. + + stream_.streamTime += ( stream_.bufferSize * 1.0 / stream_.sampleRate ); + +#if defined( HAVE_GETTIMEOFDAY ) + gettimeofday( &stream_.lastTickTimestamp, NULL ); +#endif +} + +long RtApi :: getStreamLatency( void ) +{ + verifyStream(); + + long totalLatency = 0; + if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) + totalLatency = stream_.latency[0]; + if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) + totalLatency += stream_.latency[1]; + + return totalLatency; +} + +double RtApi :: getStreamTime( void ) +{ + verifyStream(); + +#if defined( HAVE_GETTIMEOFDAY ) + // Return a very accurate estimate of the stream time by + // adding in the elapsed time since the last tick. + struct timeval then; + struct timeval now; + + if ( stream_.state != STREAM_RUNNING || stream_.streamTime == 0.0 ) + return stream_.streamTime; + + gettimeofday( &now, NULL ); + then = stream_.lastTickTimestamp; + return stream_.streamTime + + ((now.tv_sec + 0.000001 * now.tv_usec) - + (then.tv_sec + 0.000001 * then.tv_usec)); +#else + return stream_.streamTime; +#endif +} + +void RtApi :: setStreamTime( double time ) +{ + verifyStream(); + + if ( time >= 0.0 ) + stream_.streamTime = time; +#if defined( HAVE_GETTIMEOFDAY ) + gettimeofday( &stream_.lastTickTimestamp, NULL ); +#endif +} + +unsigned int RtApi :: getStreamSampleRate( void ) +{ + verifyStream(); + + return stream_.sampleRate; +} + + +// *************************************************** // +// +// OS/API-specific methods. +// +// *************************************************** // + +#if defined(__MACOSX_CORE__) + +// The OS X CoreAudio API is designed to use a separate callback +// procedure for each of its audio devices. A single RtAudio duplex +// stream using two different devices is supported here, though it +// cannot be guaranteed to always behave correctly because we cannot +// synchronize these two callbacks. +// +// A property listener is installed for over/underrun information. +// However, no functionality is currently provided to allow property +// listeners to trigger user handlers because it is unclear what could +// be done if a critical stream parameter (buffer size, sample rate, +// device disconnect) notification arrived. The listeners entail +// quite a bit of extra code and most likely, a user program wouldn't +// be prepared for the result anyway. However, we do provide a flag +// to the client callback function to inform of an over/underrun. + +// A structure to hold various information related to the CoreAudio API +// implementation. +struct CoreHandle { + AudioDeviceID id[2]; // device ids +#if defined( MAC_OS_X_VERSION_10_5 ) && ( MAC_OS_X_VERSION_MIN_REQUIRED >= MAC_OS_X_VERSION_10_5 ) + AudioDeviceIOProcID procId[2]; +#endif + UInt32 iStream[2]; // device stream index (or first if using multiple) + UInt32 nStreams[2]; // number of streams to use + bool xrun[2]; + char *deviceBuffer; + pthread_cond_t condition; + int drainCounter; // Tracks callback counts when draining + bool internalDrain; // Indicates if stop is initiated from callback or not. + + CoreHandle() + :deviceBuffer(0), drainCounter(0), internalDrain(false) { nStreams[0] = 1; nStreams[1] = 1; id[0] = 0; id[1] = 0; xrun[0] = false; xrun[1] = false; } +}; + +RtApiCore:: RtApiCore() +{ +#if defined( AVAILABLE_MAC_OS_X_VERSION_10_6_AND_LATER ) + // This is a largely undocumented but absolutely necessary + // requirement starting with OS-X 10.6. If not called, queries and + // updates to various audio device properties are not handled + // correctly. + CFRunLoopRef theRunLoop = NULL; + AudioObjectPropertyAddress property = { kAudioHardwarePropertyRunLoop, + kAudioObjectPropertyScopeGlobal, + kAudioObjectPropertyElementMaster }; + OSStatus result = AudioObjectSetPropertyData( kAudioObjectSystemObject, &property, 0, NULL, sizeof(CFRunLoopRef), &theRunLoop); + if ( result != noErr ) { + errorText_ = "RtApiCore::RtApiCore: error setting run loop property!"; + error( RtAudioError::WARNING ); + } +#endif +} + +RtApiCore :: ~RtApiCore() +{ + // The subclass destructor gets called before the base class + // destructor, so close an existing stream before deallocating + // apiDeviceId memory. + if ( stream_.state != STREAM_CLOSED ) closeStream(); +} + +unsigned int RtApiCore :: getDeviceCount( void ) +{ + // Find out how many audio devices there are, if any. + UInt32 dataSize; + AudioObjectPropertyAddress propertyAddress = { kAudioHardwarePropertyDevices, kAudioObjectPropertyScopeGlobal, kAudioObjectPropertyElementMaster }; + OSStatus result = AudioObjectGetPropertyDataSize( kAudioObjectSystemObject, &propertyAddress, 0, NULL, &dataSize ); + if ( result != noErr ) { + errorText_ = "RtApiCore::getDeviceCount: OS-X error getting device info!"; + error( RtAudioError::WARNING ); + return 0; + } + + return dataSize / sizeof( AudioDeviceID ); +} + +unsigned int RtApiCore :: getDefaultInputDevice( void ) +{ + unsigned int nDevices = getDeviceCount(); + if ( nDevices <= 1 ) return 0; + + AudioDeviceID id; + UInt32 dataSize = sizeof( AudioDeviceID ); + AudioObjectPropertyAddress property = { kAudioHardwarePropertyDefaultInputDevice, kAudioObjectPropertyScopeGlobal, kAudioObjectPropertyElementMaster }; + OSStatus result = AudioObjectGetPropertyData( kAudioObjectSystemObject, &property, 0, NULL, &dataSize, &id ); + if ( result != noErr ) { + errorText_ = "RtApiCore::getDefaultInputDevice: OS-X system error getting device."; + error( RtAudioError::WARNING ); + return 0; + } + + dataSize *= nDevices; + AudioDeviceID deviceList[ nDevices ]; + property.mSelector = kAudioHardwarePropertyDevices; + result = AudioObjectGetPropertyData( kAudioObjectSystemObject, &property, 0, NULL, &dataSize, (void *) &deviceList ); + if ( result != noErr ) { + errorText_ = "RtApiCore::getDefaultInputDevice: OS-X system error getting device IDs."; + error( RtAudioError::WARNING ); + return 0; + } + + for ( unsigned int i=0; i<nDevices; i++ ) + if ( id == deviceList[i] ) return i; + + errorText_ = "RtApiCore::getDefaultInputDevice: No default device found!"; + error( RtAudioError::WARNING ); + return 0; +} + +unsigned int RtApiCore :: getDefaultOutputDevice( void ) +{ + unsigned int nDevices = getDeviceCount(); + if ( nDevices <= 1 ) return 0; + + AudioDeviceID id; + UInt32 dataSize = sizeof( AudioDeviceID ); + AudioObjectPropertyAddress property = { kAudioHardwarePropertyDefaultOutputDevice, kAudioObjectPropertyScopeGlobal, kAudioObjectPropertyElementMaster }; + OSStatus result = AudioObjectGetPropertyData( kAudioObjectSystemObject, &property, 0, NULL, &dataSize, &id ); + if ( result != noErr ) { + errorText_ = "RtApiCore::getDefaultOutputDevice: OS-X system error getting device."; + error( RtAudioError::WARNING ); + return 0; + } + + dataSize = sizeof( AudioDeviceID ) * nDevices; + AudioDeviceID deviceList[ nDevices ]; + property.mSelector = kAudioHardwarePropertyDevices; + result = AudioObjectGetPropertyData( kAudioObjectSystemObject, &property, 0, NULL, &dataSize, (void *) &deviceList ); + if ( result != noErr ) { + errorText_ = "RtApiCore::getDefaultOutputDevice: OS-X system error getting device IDs."; + error( RtAudioError::WARNING ); + return 0; + } + + for ( unsigned int i=0; i<nDevices; i++ ) + if ( id == deviceList[i] ) return i; + + errorText_ = "RtApiCore::getDefaultOutputDevice: No default device found!"; + error( RtAudioError::WARNING ); + return 0; +} + +RtAudio::DeviceInfo RtApiCore :: getDeviceInfo( unsigned int device ) +{ + RtAudio::DeviceInfo info; + info.probed = false; + + // Get device ID + unsigned int nDevices = getDeviceCount(); + if ( nDevices == 0 ) { + errorText_ = "RtApiCore::getDeviceInfo: no devices found!"; + error( RtAudioError::INVALID_USE ); + return info; + } + + if ( device >= nDevices ) { + errorText_ = "RtApiCore::getDeviceInfo: device ID is invalid!"; + error( RtAudioError::INVALID_USE ); + return info; + } + + AudioDeviceID deviceList[ nDevices ]; + UInt32 dataSize = sizeof( AudioDeviceID ) * nDevices; + AudioObjectPropertyAddress property = { kAudioHardwarePropertyDevices, + kAudioObjectPropertyScopeGlobal, + kAudioObjectPropertyElementMaster }; + OSStatus result = AudioObjectGetPropertyData( kAudioObjectSystemObject, &property, + 0, NULL, &dataSize, (void *) &deviceList ); + if ( result != noErr ) { + errorText_ = "RtApiCore::getDeviceInfo: OS-X system error getting device IDs."; + error( RtAudioError::WARNING ); + return info; + } + + AudioDeviceID id = deviceList[ device ]; + + // Get the device name. + info.name.erase(); + CFStringRef cfname; + dataSize = sizeof( CFStringRef ); + property.mSelector = kAudioObjectPropertyManufacturer; + result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, &cfname ); + if ( result != noErr ) { + errorStream_ << "RtApiCore::probeDeviceInfo: system error (" << getErrorCode( result ) << ") getting device manufacturer."; + errorText_ = errorStream_.str(); + error( RtAudioError::WARNING ); + return info; + } + + //const char *mname = CFStringGetCStringPtr( cfname, CFStringGetSystemEncoding() ); + int length = CFStringGetLength(cfname); + char *mname = (char *)malloc(length * 3 + 1); +#if defined( UNICODE ) || defined( _UNICODE ) + CFStringGetCString(cfname, mname, length * 3 + 1, kCFStringEncodingUTF8); +#else + CFStringGetCString(cfname, mname, length * 3 + 1, CFStringGetSystemEncoding()); +#endif + info.name.append( (const char *)mname, strlen(mname) ); + info.name.append( ": " ); + CFRelease( cfname ); + free(mname); + + property.mSelector = kAudioObjectPropertyName; + result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, &cfname ); + if ( result != noErr ) { + errorStream_ << "RtApiCore::probeDeviceInfo: system error (" << getErrorCode( result ) << ") getting device name."; + errorText_ = errorStream_.str(); + error( RtAudioError::WARNING ); + return info; + } + + //const char *name = CFStringGetCStringPtr( cfname, CFStringGetSystemEncoding() ); + length = CFStringGetLength(cfname); + char *name = (char *)malloc(length * 3 + 1); +#if defined( UNICODE ) || defined( _UNICODE ) + CFStringGetCString(cfname, name, length * 3 + 1, kCFStringEncodingUTF8); +#else + CFStringGetCString(cfname, name, length * 3 + 1, CFStringGetSystemEncoding()); +#endif + info.name.append( (const char *)name, strlen(name) ); + CFRelease( cfname ); + free(name); + + // Get the output stream "configuration". + AudioBufferList *bufferList = nil; + property.mSelector = kAudioDevicePropertyStreamConfiguration; + property.mScope = kAudioDevicePropertyScopeOutput; + // property.mElement = kAudioObjectPropertyElementWildcard; + dataSize = 0; + result = AudioObjectGetPropertyDataSize( id, &property, 0, NULL, &dataSize ); + if ( result != noErr || dataSize == 0 ) { + errorStream_ << "RtApiCore::getDeviceInfo: system error (" << getErrorCode( result ) << ") getting output stream configuration info for device (" << device << ")."; + errorText_ = errorStream_.str(); + error( RtAudioError::WARNING ); + return info; + } + + // Allocate the AudioBufferList. + bufferList = (AudioBufferList *) malloc( dataSize ); + if ( bufferList == NULL ) { + errorText_ = "RtApiCore::getDeviceInfo: memory error allocating output AudioBufferList."; + error( RtAudioError::WARNING ); + return info; + } + + result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, bufferList ); + if ( result != noErr || dataSize == 0 ) { + free( bufferList ); + errorStream_ << "RtApiCore::getDeviceInfo: system error (" << getErrorCode( result ) << ") getting output stream configuration for device (" << device << ")."; + errorText_ = errorStream_.str(); + error( RtAudioError::WARNING ); + return info; + } + + // Get output channel information. + unsigned int i, nStreams = bufferList->mNumberBuffers; + for ( i=0; i<nStreams; i++ ) + info.outputChannels += bufferList->mBuffers[i].mNumberChannels; + free( bufferList ); + + // Get the input stream "configuration". + property.mScope = kAudioDevicePropertyScopeInput; + result = AudioObjectGetPropertyDataSize( id, &property, 0, NULL, &dataSize ); + if ( result != noErr || dataSize == 0 ) { + errorStream_ << "RtApiCore::getDeviceInfo: system error (" << getErrorCode( result ) << ") getting input stream configuration info for device (" << device << ")."; + errorText_ = errorStream_.str(); + error( RtAudioError::WARNING ); + return info; + } + + // Allocate the AudioBufferList. + bufferList = (AudioBufferList *) malloc( dataSize ); + if ( bufferList == NULL ) { + errorText_ = "RtApiCore::getDeviceInfo: memory error allocating input AudioBufferList."; + error( RtAudioError::WARNING ); + return info; + } + + result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, bufferList ); + if (result != noErr || dataSize == 0) { + free( bufferList ); + errorStream_ << "RtApiCore::getDeviceInfo: system error (" << getErrorCode( result ) << ") getting input stream configuration for device (" << device << ")."; + errorText_ = errorStream_.str(); + error( RtAudioError::WARNING ); + return info; + } + + // Get input channel information. + nStreams = bufferList->mNumberBuffers; + for ( i=0; i<nStreams; i++ ) + info.inputChannels += bufferList->mBuffers[i].mNumberChannels; + free( bufferList ); + + // If device opens for both playback and capture, we determine the channels. + if ( info.outputChannels > 0 && info.inputChannels > 0 ) + info.duplexChannels = (info.outputChannels > info.inputChannels) ? info.inputChannels : info.outputChannels; + + // Probe the device sample rates. + bool isInput = false; + if ( info.outputChannels == 0 ) isInput = true; + + // Determine the supported sample rates. + property.mSelector = kAudioDevicePropertyAvailableNominalSampleRates; + if ( isInput == false ) property.mScope = kAudioDevicePropertyScopeOutput; + result = AudioObjectGetPropertyDataSize( id, &property, 0, NULL, &dataSize ); + if ( result != kAudioHardwareNoError || dataSize == 0 ) { + errorStream_ << "RtApiCore::getDeviceInfo: system error (" << getErrorCode( result ) << ") getting sample rate info."; + errorText_ = errorStream_.str(); + error( RtAudioError::WARNING ); + return info; + } + + UInt32 nRanges = dataSize / sizeof( AudioValueRange ); + AudioValueRange rangeList[ nRanges ]; + result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, &rangeList ); + if ( result != kAudioHardwareNoError ) { + errorStream_ << "RtApiCore::getDeviceInfo: system error (" << getErrorCode( result ) << ") getting sample rates."; + errorText_ = errorStream_.str(); + error( RtAudioError::WARNING ); + return info; + } + + // The sample rate reporting mechanism is a bit of a mystery. It + // seems that it can either return individual rates or a range of + // rates. I assume that if the min / max range values are the same, + // then that represents a single supported rate and if the min / max + // range values are different, the device supports an arbitrary + // range of values (though there might be multiple ranges, so we'll + // use the most conservative range). + Float64 minimumRate = 1.0, maximumRate = 10000000000.0; + bool haveValueRange = false; + info.sampleRates.clear(); + for ( UInt32 i=0; i<nRanges; i++ ) { + if ( rangeList[i].mMinimum == rangeList[i].mMaximum ) { + unsigned int tmpSr = (unsigned int) rangeList[i].mMinimum; + info.sampleRates.push_back( tmpSr ); + + if ( !info.preferredSampleRate || ( tmpSr <= 48000 && tmpSr > info.preferredSampleRate ) ) + info.preferredSampleRate = tmpSr; + + } else { + haveValueRange = true; + if ( rangeList[i].mMinimum > minimumRate ) minimumRate = rangeList[i].mMinimum; + if ( rangeList[i].mMaximum < maximumRate ) maximumRate = rangeList[i].mMaximum; + } + } + + if ( haveValueRange ) { + for ( unsigned int k=0; k<MAX_SAMPLE_RATES; k++ ) { + if ( SAMPLE_RATES[k] >= (unsigned int) minimumRate && SAMPLE_RATES[k] <= (unsigned int) maximumRate ) { + info.sampleRates.push_back( SAMPLE_RATES[k] ); + + if ( !info.preferredSampleRate || ( SAMPLE_RATES[k] <= 48000 && SAMPLE_RATES[k] > info.preferredSampleRate ) ) + info.preferredSampleRate = SAMPLE_RATES[k]; + } + } + } + + // Sort and remove any redundant values + std::sort( info.sampleRates.begin(), info.sampleRates.end() ); + info.sampleRates.erase( unique( info.sampleRates.begin(), info.sampleRates.end() ), info.sampleRates.end() ); + + if ( info.sampleRates.size() == 0 ) { + errorStream_ << "RtApiCore::probeDeviceInfo: No supported sample rates found for device (" << device << ")."; + errorText_ = errorStream_.str(); + error( RtAudioError::WARNING ); + return info; + } + + // CoreAudio always uses 32-bit floating point data for PCM streams. + // Thus, any other "physical" formats supported by the device are of + // no interest to the client. + info.nativeFormats = RTAUDIO_FLOAT32; + + if ( info.outputChannels > 0 ) + if ( getDefaultOutputDevice() == device ) info.isDefaultOutput = true; + if ( info.inputChannels > 0 ) + if ( getDefaultInputDevice() == device ) info.isDefaultInput = true; + + info.probed = true; + return info; +} + +static OSStatus callbackHandler( AudioDeviceID inDevice, + const AudioTimeStamp* /*inNow*/, + const AudioBufferList* inInputData, + const AudioTimeStamp* /*inInputTime*/, + AudioBufferList* outOutputData, + const AudioTimeStamp* /*inOutputTime*/, + void* infoPointer ) +{ + CallbackInfo *info = (CallbackInfo *) infoPointer; + + RtApiCore *object = (RtApiCore *) info->object; + if ( object->callbackEvent( inDevice, inInputData, outOutputData ) == false ) + return kAudioHardwareUnspecifiedError; + else + return kAudioHardwareNoError; +} + +static OSStatus xrunListener( AudioObjectID /*inDevice*/, + UInt32 nAddresses, + const AudioObjectPropertyAddress properties[], + void* handlePointer ) +{ + CoreHandle *handle = (CoreHandle *) handlePointer; + for ( UInt32 i=0; i<nAddresses; i++ ) { + if ( properties[i].mSelector == kAudioDeviceProcessorOverload ) { + if ( properties[i].mScope == kAudioDevicePropertyScopeInput ) + handle->xrun[1] = true; + else + handle->xrun[0] = true; + } + } + + return kAudioHardwareNoError; +} + +static OSStatus rateListener( AudioObjectID inDevice, + UInt32 /*nAddresses*/, + const AudioObjectPropertyAddress /*properties*/[], + void* ratePointer ) +{ + Float64 *rate = (Float64 *) ratePointer; + UInt32 dataSize = sizeof( Float64 ); + AudioObjectPropertyAddress property = { kAudioDevicePropertyNominalSampleRate, + kAudioObjectPropertyScopeGlobal, + kAudioObjectPropertyElementMaster }; + AudioObjectGetPropertyData( inDevice, &property, 0, NULL, &dataSize, rate ); + return kAudioHardwareNoError; +} + +bool RtApiCore :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels, + unsigned int firstChannel, unsigned int sampleRate, + RtAudioFormat format, unsigned int *bufferSize, + RtAudio::StreamOptions *options ) +{ + // Get device ID + unsigned int nDevices = getDeviceCount(); + if ( nDevices == 0 ) { + // This should not happen because a check is made before this function is called. + errorText_ = "RtApiCore::probeDeviceOpen: no devices found!"; + return FAILURE; + } + + if ( device >= nDevices ) { + // This should not happen because a check is made before this function is called. + errorText_ = "RtApiCore::probeDeviceOpen: device ID is invalid!"; + return FAILURE; + } + + AudioDeviceID deviceList[ nDevices ]; + UInt32 dataSize = sizeof( AudioDeviceID ) * nDevices; + AudioObjectPropertyAddress property = { kAudioHardwarePropertyDevices, + kAudioObjectPropertyScopeGlobal, + kAudioObjectPropertyElementMaster }; + OSStatus result = AudioObjectGetPropertyData( kAudioObjectSystemObject, &property, + 0, NULL, &dataSize, (void *) &deviceList ); + if ( result != noErr ) { + errorText_ = "RtApiCore::probeDeviceOpen: OS-X system error getting device IDs."; + return FAILURE; + } + + AudioDeviceID id = deviceList[ device ]; + + // Setup for stream mode. + bool isInput = false; + if ( mode == INPUT ) { + isInput = true; + property.mScope = kAudioDevicePropertyScopeInput; + } + else + property.mScope = kAudioDevicePropertyScopeOutput; + + // Get the stream "configuration". + AudioBufferList *bufferList = nil; + dataSize = 0; + property.mSelector = kAudioDevicePropertyStreamConfiguration; + result = AudioObjectGetPropertyDataSize( id, &property, 0, NULL, &dataSize ); + if ( result != noErr || dataSize == 0 ) { + errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting stream configuration info for device (" << device << ")."; + errorText_ = errorStream_.str(); + return FAILURE; + } + + // Allocate the AudioBufferList. + bufferList = (AudioBufferList *) malloc( dataSize ); + if ( bufferList == NULL ) { + errorText_ = "RtApiCore::probeDeviceOpen: memory error allocating AudioBufferList."; + return FAILURE; + } + + result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, bufferList ); + if (result != noErr || dataSize == 0) { + free( bufferList ); + errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting stream configuration for device (" << device << ")."; + errorText_ = errorStream_.str(); + return FAILURE; + } + + // Search for one or more streams that contain the desired number of + // channels. CoreAudio devices can have an arbitrary number of + // streams and each stream can have an arbitrary number of channels. + // For each stream, a single buffer of interleaved samples is + // provided. RtAudio prefers the use of one stream of interleaved + // data or multiple consecutive single-channel streams. However, we + // now support multiple consecutive multi-channel streams of + // interleaved data as well. + UInt32 iStream, offsetCounter = firstChannel; + UInt32 nStreams = bufferList->mNumberBuffers; + bool monoMode = false; + bool foundStream = false; + + // First check that the device supports the requested number of + // channels. + UInt32 deviceChannels = 0; + for ( iStream=0; iStream<nStreams; iStream++ ) + deviceChannels += bufferList->mBuffers[iStream].mNumberChannels; + + if ( deviceChannels < ( channels + firstChannel ) ) { + free( bufferList ); + errorStream_ << "RtApiCore::probeDeviceOpen: the device (" << device << ") does not support the requested channel count."; + errorText_ = errorStream_.str(); + return FAILURE; + } + + // Look for a single stream meeting our needs. + UInt32 firstStream, streamCount = 1, streamChannels = 0, channelOffset = 0; + for ( iStream=0; iStream<nStreams; iStream++ ) { + streamChannels = bufferList->mBuffers[iStream].mNumberChannels; + if ( streamChannels >= channels + offsetCounter ) { + firstStream = iStream; + channelOffset = offsetCounter; + foundStream = true; + break; + } + if ( streamChannels > offsetCounter ) break; + offsetCounter -= streamChannels; + } + + // If we didn't find a single stream above, then we should be able + // to meet the channel specification with multiple streams. + if ( foundStream == false ) { + monoMode = true; + offsetCounter = firstChannel; + for ( iStream=0; iStream<nStreams; iStream++ ) { + streamChannels = bufferList->mBuffers[iStream].mNumberChannels; + if ( streamChannels > offsetCounter ) break; + offsetCounter -= streamChannels; + } + + firstStream = iStream; + channelOffset = offsetCounter; + Int32 channelCounter = channels + offsetCounter - streamChannels; + + if ( streamChannels > 1 ) monoMode = false; + while ( channelCounter > 0 ) { + streamChannels = bufferList->mBuffers[++iStream].mNumberChannels; + if ( streamChannels > 1 ) monoMode = false; + channelCounter -= streamChannels; + streamCount++; + } + } + + free( bufferList ); + + // Determine the buffer size. + AudioValueRange bufferRange; + dataSize = sizeof( AudioValueRange ); + property.mSelector = kAudioDevicePropertyBufferFrameSizeRange; + result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, &bufferRange ); + + if ( result != noErr ) { + errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting buffer size range for device (" << device << ")."; + errorText_ = errorStream_.str(); + return FAILURE; + } + + if ( bufferRange.mMinimum > *bufferSize ) *bufferSize = (unsigned long) bufferRange.mMinimum; + else if ( bufferRange.mMaximum < *bufferSize ) *bufferSize = (unsigned long) bufferRange.mMaximum; + if ( options && options->flags & RTAUDIO_MINIMIZE_LATENCY ) *bufferSize = (unsigned long) bufferRange.mMinimum; + + // Set the buffer size. For multiple streams, I'm assuming we only + // need to make this setting for the master channel. + UInt32 theSize = (UInt32) *bufferSize; + dataSize = sizeof( UInt32 ); + property.mSelector = kAudioDevicePropertyBufferFrameSize; + result = AudioObjectSetPropertyData( id, &property, 0, NULL, dataSize, &theSize ); + + if ( result != noErr ) { + errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") setting the buffer size for device (" << device << ")."; + errorText_ = errorStream_.str(); + return FAILURE; + } + + // If attempting to setup a duplex stream, the bufferSize parameter + // MUST be the same in both directions! + *bufferSize = theSize; + if ( stream_.mode == OUTPUT && mode == INPUT && *bufferSize != stream_.bufferSize ) { + errorStream_ << "RtApiCore::probeDeviceOpen: system error setting buffer size for duplex stream on device (" << device << ")."; + errorText_ = errorStream_.str(); + return FAILURE; + } + + stream_.bufferSize = *bufferSize; + stream_.nBuffers = 1; + + // Try to set "hog" mode ... it's not clear to me this is working. + if ( options && options->flags & RTAUDIO_HOG_DEVICE ) { + pid_t hog_pid; + dataSize = sizeof( hog_pid ); + property.mSelector = kAudioDevicePropertyHogMode; + result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, &hog_pid ); + if ( result != noErr ) { + errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting 'hog' state!"; + errorText_ = errorStream_.str(); + return FAILURE; + } + + if ( hog_pid != getpid() ) { + hog_pid = getpid(); + result = AudioObjectSetPropertyData( id, &property, 0, NULL, dataSize, &hog_pid ); + if ( result != noErr ) { + errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") setting 'hog' state!"; + errorText_ = errorStream_.str(); + return FAILURE; + } + } + } + + // Check and if necessary, change the sample rate for the device. + Float64 nominalRate; + dataSize = sizeof( Float64 ); + property.mSelector = kAudioDevicePropertyNominalSampleRate; + result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, &nominalRate ); + if ( result != noErr ) { + errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting current sample rate."; + errorText_ = errorStream_.str(); + return FAILURE; + } + + // Only change the sample rate if off by more than 1 Hz. + if ( fabs( nominalRate - (double)sampleRate ) > 1.0 ) { + + // Set a property listener for the sample rate change + Float64 reportedRate = 0.0; + AudioObjectPropertyAddress tmp = { kAudioDevicePropertyNominalSampleRate, kAudioObjectPropertyScopeGlobal, kAudioObjectPropertyElementMaster }; + result = AudioObjectAddPropertyListener( id, &tmp, rateListener, (void *) &reportedRate ); + if ( result != noErr ) { + errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") setting sample rate property listener for device (" << device << ")."; + errorText_ = errorStream_.str(); + return FAILURE; + } + + nominalRate = (Float64) sampleRate; + result = AudioObjectSetPropertyData( id, &property, 0, NULL, dataSize, &nominalRate ); + if ( result != noErr ) { + AudioObjectRemovePropertyListener( id, &tmp, rateListener, (void *) &reportedRate ); + errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") setting sample rate for device (" << device << ")."; + errorText_ = errorStream_.str(); + return FAILURE; + } + + // Now wait until the reported nominal rate is what we just set. + UInt32 microCounter = 0; + while ( reportedRate != nominalRate ) { + microCounter += 5000; + if ( microCounter > 5000000 ) break; + usleep( 5000 ); + } + + // Remove the property listener. + AudioObjectRemovePropertyListener( id, &tmp, rateListener, (void *) &reportedRate ); + + if ( microCounter > 5000000 ) { + errorStream_ << "RtApiCore::probeDeviceOpen: timeout waiting for sample rate update for device (" << device << ")."; + errorText_ = errorStream_.str(); + return FAILURE; + } + } + + // Now set the stream format for all streams. Also, check the + // physical format of the device and change that if necessary. + AudioStreamBasicDescription description; + dataSize = sizeof( AudioStreamBasicDescription ); + property.mSelector = kAudioStreamPropertyVirtualFormat; + result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, &description ); + if ( result != noErr ) { + errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting stream format for device (" << device << ")."; + errorText_ = errorStream_.str(); + return FAILURE; + } + + // Set the sample rate and data format id. However, only make the + // change if the sample rate is not within 1.0 of the desired + // rate and the format is not linear pcm. + bool updateFormat = false; + if ( fabs( description.mSampleRate - (Float64)sampleRate ) > 1.0 ) { + description.mSampleRate = (Float64) sampleRate; + updateFormat = true; + } + + if ( description.mFormatID != kAudioFormatLinearPCM ) { + description.mFormatID = kAudioFormatLinearPCM; + updateFormat = true; + } + + if ( updateFormat ) { + result = AudioObjectSetPropertyData( id, &property, 0, NULL, dataSize, &description ); + if ( result != noErr ) { + errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") setting sample rate or data format for device (" << device << ")."; + errorText_ = errorStream_.str(); + return FAILURE; + } + } + + // Now check the physical format. + property.mSelector = kAudioStreamPropertyPhysicalFormat; + result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, &description ); + if ( result != noErr ) { + errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting stream physical format for device (" << device << ")."; + errorText_ = errorStream_.str(); + return FAILURE; + } + + //std::cout << "Current physical stream format:" << std::endl; + //std::cout << " mBitsPerChan = " << description.mBitsPerChannel << std::endl; + //std::cout << " aligned high = " << (description.mFormatFlags & kAudioFormatFlagIsAlignedHigh) << ", isPacked = " << (description.mFormatFlags & kAudioFormatFlagIsPacked) << std::endl; + //std::cout << " bytesPerFrame = " << description.mBytesPerFrame << std::endl; + //std::cout << " sample rate = " << description.mSampleRate << std::endl; + + if ( description.mFormatID != kAudioFormatLinearPCM || description.mBitsPerChannel < 16 ) { + description.mFormatID = kAudioFormatLinearPCM; + //description.mSampleRate = (Float64) sampleRate; + AudioStreamBasicDescription testDescription = description; + UInt32 formatFlags; + + // We'll try higher bit rates first and then work our way down. + std::vector< std::pair<UInt32, UInt32> > physicalFormats; + formatFlags = (description.mFormatFlags | kLinearPCMFormatFlagIsFloat) & ~kLinearPCMFormatFlagIsSignedInteger; + physicalFormats.push_back( std::pair<Float32, UInt32>( 32, formatFlags ) ); + formatFlags = (description.mFormatFlags | kLinearPCMFormatFlagIsSignedInteger | kAudioFormatFlagIsPacked) & ~kLinearPCMFormatFlagIsFloat; + physicalFormats.push_back( std::pair<Float32, UInt32>( 32, formatFlags ) ); + physicalFormats.push_back( std::pair<Float32, UInt32>( 24, formatFlags ) ); // 24-bit packed + formatFlags &= ~( kAudioFormatFlagIsPacked | kAudioFormatFlagIsAlignedHigh ); + physicalFormats.push_back( std::pair<Float32, UInt32>( 24.2, formatFlags ) ); // 24-bit in 4 bytes, aligned low + formatFlags |= kAudioFormatFlagIsAlignedHigh; + physicalFormats.push_back( std::pair<Float32, UInt32>( 24.4, formatFlags ) ); // 24-bit in 4 bytes, aligned high + formatFlags = (description.mFormatFlags | kLinearPCMFormatFlagIsSignedInteger | kAudioFormatFlagIsPacked) & ~kLinearPCMFormatFlagIsFloat; + physicalFormats.push_back( std::pair<Float32, UInt32>( 16, formatFlags ) ); + physicalFormats.push_back( std::pair<Float32, UInt32>( 8, formatFlags ) ); + + bool setPhysicalFormat = false; + for( unsigned int i=0; i<physicalFormats.size(); i++ ) { + testDescription = description; + testDescription.mBitsPerChannel = (UInt32) physicalFormats[i].first; + testDescription.mFormatFlags = physicalFormats[i].second; + if ( (24 == (UInt32)physicalFormats[i].first) && ~( physicalFormats[i].second & kAudioFormatFlagIsPacked ) ) + testDescription.mBytesPerFrame = 4 * testDescription.mChannelsPerFrame; + else + testDescription.mBytesPerFrame = testDescription.mBitsPerChannel/8 * testDescription.mChannelsPerFrame; + testDescription.mBytesPerPacket = testDescription.mBytesPerFrame * testDescription.mFramesPerPacket; + result = AudioObjectSetPropertyData( id, &property, 0, NULL, dataSize, &testDescription ); + if ( result == noErr ) { + setPhysicalFormat = true; + //std::cout << "Updated physical stream format:" << std::endl; + //std::cout << " mBitsPerChan = " << testDescription.mBitsPerChannel << std::endl; + //std::cout << " aligned high = " << (testDescription.mFormatFlags & kAudioFormatFlagIsAlignedHigh) << ", isPacked = " << (testDescription.mFormatFlags & kAudioFormatFlagIsPacked) << std::endl; + //std::cout << " bytesPerFrame = " << testDescription.mBytesPerFrame << std::endl; + //std::cout << " sample rate = " << testDescription.mSampleRate << std::endl; + break; + } + } + + if ( !setPhysicalFormat ) { + errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") setting physical data format for device (" << device << ")."; + errorText_ = errorStream_.str(); + return FAILURE; + } + } // done setting virtual/physical formats. + + // Get the stream / device latency. + UInt32 latency; + dataSize = sizeof( UInt32 ); + property.mSelector = kAudioDevicePropertyLatency; + if ( AudioObjectHasProperty( id, &property ) == true ) { + result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, &latency ); + if ( result == kAudioHardwareNoError ) stream_.latency[ mode ] = latency; + else { + errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting device latency for device (" << device << ")."; + errorText_ = errorStream_.str(); + error( RtAudioError::WARNING ); + } + } + + // Byte-swapping: According to AudioHardware.h, the stream data will + // always be presented in native-endian format, so we should never + // need to byte swap. + stream_.doByteSwap[mode] = false; + + // From the CoreAudio documentation, PCM data must be supplied as + // 32-bit floats. + stream_.userFormat = format; + stream_.deviceFormat[mode] = RTAUDIO_FLOAT32; + + if ( streamCount == 1 ) + stream_.nDeviceChannels[mode] = description.mChannelsPerFrame; + else // multiple streams + stream_.nDeviceChannels[mode] = channels; + stream_.nUserChannels[mode] = channels; + stream_.channelOffset[mode] = channelOffset; // offset within a CoreAudio stream + if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) stream_.userInterleaved = false; + else stream_.userInterleaved = true; + stream_.deviceInterleaved[mode] = true; + if ( monoMode == true ) stream_.deviceInterleaved[mode] = false; + + // Set flags for buffer conversion. + stream_.doConvertBuffer[mode] = false; + if ( stream_.userFormat != stream_.deviceFormat[mode] ) + stream_.doConvertBuffer[mode] = true; + if ( stream_.nUserChannels[mode] < stream_.nDeviceChannels[mode] ) + stream_.doConvertBuffer[mode] = true; + if ( streamCount == 1 ) { + if ( stream_.nUserChannels[mode] > 1 && + stream_.userInterleaved != stream_.deviceInterleaved[mode] ) + stream_.doConvertBuffer[mode] = true; + } + else if ( monoMode && stream_.userInterleaved ) + stream_.doConvertBuffer[mode] = true; + + // Allocate our CoreHandle structure for the stream. + CoreHandle *handle = 0; + if ( stream_.apiHandle == 0 ) { + try { + handle = new CoreHandle; + } + catch ( std::bad_alloc& ) { + errorText_ = "RtApiCore::probeDeviceOpen: error allocating CoreHandle memory."; + goto error; + } + + if ( pthread_cond_init( &handle->condition, NULL ) ) { + errorText_ = "RtApiCore::probeDeviceOpen: error initializing pthread condition variable."; + goto error; + } + stream_.apiHandle = (void *) handle; + } + else + handle = (CoreHandle *) stream_.apiHandle; + handle->iStream[mode] = firstStream; + handle->nStreams[mode] = streamCount; + handle->id[mode] = id; + + // Allocate necessary internal buffers. + unsigned long bufferBytes; + bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat ); + // stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 ); + stream_.userBuffer[mode] = (char *) malloc( bufferBytes * sizeof(char) ); + memset( stream_.userBuffer[mode], 0, bufferBytes * sizeof(char) ); + if ( stream_.userBuffer[mode] == NULL ) { + errorText_ = "RtApiCore::probeDeviceOpen: error allocating user buffer memory."; + goto error; + } + + // If possible, we will make use of the CoreAudio stream buffers as + // "device buffers". However, we can't do this if using multiple + // streams. + if ( stream_.doConvertBuffer[mode] && handle->nStreams[mode] > 1 ) { + + bool makeBuffer = true; + bufferBytes = stream_.nDeviceChannels[mode] * formatBytes( stream_.deviceFormat[mode] ); + if ( mode == INPUT ) { + if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) { + unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] ); + if ( bufferBytes <= bytesOut ) makeBuffer = false; + } + } + + if ( makeBuffer ) { + bufferBytes *= *bufferSize; + if ( stream_.deviceBuffer ) free( stream_.deviceBuffer ); + stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 ); + if ( stream_.deviceBuffer == NULL ) { + errorText_ = "RtApiCore::probeDeviceOpen: error allocating device buffer memory."; + goto error; + } + } + } + + stream_.sampleRate = sampleRate; + stream_.device[mode] = device; + stream_.state = STREAM_STOPPED; + stream_.callbackInfo.object = (void *) this; + + // Setup the buffer conversion information structure. + if ( stream_.doConvertBuffer[mode] ) { + if ( streamCount > 1 ) setConvertInfo( mode, 0 ); + else setConvertInfo( mode, channelOffset ); + } + + if ( mode == INPUT && stream_.mode == OUTPUT && stream_.device[0] == device ) + // Only one callback procedure per device. + stream_.mode = DUPLEX; + else { +#if defined( MAC_OS_X_VERSION_10_5 ) && ( MAC_OS_X_VERSION_MIN_REQUIRED >= MAC_OS_X_VERSION_10_5 ) + result = AudioDeviceCreateIOProcID( id, callbackHandler, (void *) &stream_.callbackInfo, &handle->procId[mode] ); +#else + // deprecated in favor of AudioDeviceCreateIOProcID() + result = AudioDeviceAddIOProc( id, callbackHandler, (void *) &stream_.callbackInfo ); +#endif + if ( result != noErr ) { + errorStream_ << "RtApiCore::probeDeviceOpen: system error setting callback for device (" << device << ")."; + errorText_ = errorStream_.str(); + goto error; + } + if ( stream_.mode == OUTPUT && mode == INPUT ) + stream_.mode = DUPLEX; + else + stream_.mode = mode; + } + + // Setup the device property listener for over/underload. + property.mSelector = kAudioDeviceProcessorOverload; + property.mScope = kAudioObjectPropertyScopeGlobal; + result = AudioObjectAddPropertyListener( id, &property, xrunListener, (void *) handle ); + + return SUCCESS; + + error: + if ( handle ) { + pthread_cond_destroy( &handle->condition ); + delete handle; + stream_.apiHandle = 0; + } + + for ( int i=0; i<2; i++ ) { + if ( stream_.userBuffer[i] ) { + free( stream_.userBuffer[i] ); + stream_.userBuffer[i] = 0; + } + } + + if ( stream_.deviceBuffer ) { + free( stream_.deviceBuffer ); + stream_.deviceBuffer = 0; + } + + stream_.state = STREAM_CLOSED; + return FAILURE; +} + +void RtApiCore :: closeStream( void ) +{ + if ( stream_.state == STREAM_CLOSED ) { + errorText_ = "RtApiCore::closeStream(): no open stream to close!"; + error( RtAudioError::WARNING ); + return; + } + + CoreHandle *handle = (CoreHandle *) stream_.apiHandle; + if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { + if (handle) { + AudioObjectPropertyAddress property = { kAudioHardwarePropertyDevices, + kAudioObjectPropertyScopeGlobal, + kAudioObjectPropertyElementMaster }; + + property.mSelector = kAudioDeviceProcessorOverload; + property.mScope = kAudioObjectPropertyScopeGlobal; + if (AudioObjectRemovePropertyListener( handle->id[0], &property, xrunListener, (void *) handle ) != noErr) { + errorText_ = "RtApiCore::closeStream(): error removing property listener!"; + error( RtAudioError::WARNING ); + } + +#if defined( MAC_OS_X_VERSION_10_5 ) && ( MAC_OS_X_VERSION_MIN_REQUIRED >= MAC_OS_X_VERSION_10_5 ) + 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_.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 defined( MAC_OS_X_VERSION_10_5 ) && ( MAC_OS_X_VERSION_MIN_REQUIRED >= MAC_OS_X_VERSION_10_5 ) + 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_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; + } + +#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_.device[0] << ")."; + errorText_ = errorStream_.str(); + goto unlock; + } + } + + if ( stream_.mode == INPUT || + ( stream_.mode == DUPLEX && stream_.device[0] != stream_.device[1] ) ) { + +#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_.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 + } + +#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_.device[0] << ")."; + errorText_ = errorStream_.str(); + goto unlock; + } + } + + if ( stream_.mode == INPUT || ( stream_.mode == DUPLEX && stream_.device[0] != stream_.device[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_.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 ); + + // Make sure to only tick duplex stream time once if using two devices + if ( stream_.mode != DUPLEX || (stream_.mode == DUPLEX && handle->id[0] != handle->id[1] && deviceId == handle->id[0] ) ) + RtApi::tickStreamTime(); + + return SUCCESS; +} + +const char* RtApiCore :: getErrorCode( OSStatus code ) +{ + switch( code ) { + + case kAudioHardwareNotRunningError: + return "kAudioHardwareNotRunningError"; + + case kAudioHardwareUnspecifiedError: + return "kAudioHardwareUnspecifiedError"; + + case kAudioHardwareUnknownPropertyError: + return "kAudioHardwareUnknownPropertyError"; + + case kAudioHardwareBadPropertySizeError: + return "kAudioHardwareBadPropertySizeError"; + + case kAudioHardwareIllegalOperationError: + return "kAudioHardwareIllegalOperationError"; + + case kAudioHardwareBadObjectError: + return "kAudioHardwareBadObjectError"; + + case kAudioHardwareBadDeviceError: + return "kAudioHardwareBadDeviceError"; + + case kAudioHardwareBadStreamError: + return "kAudioHardwareBadStreamError"; + + case kAudioHardwareUnsupportedOperationError: + return "kAudioHardwareUnsupportedOperationError"; + + case kAudioDeviceUnsupportedFormatError: + return "kAudioDeviceUnsupportedFormatError"; + + case kAudioDevicePermissionsError: + return "kAudioDevicePermissionsError"; + + default: + return "CoreAudio unknown error"; + } +} + + //******************** End of __MACOSX_CORE__ *********************// +#endif + +#if defined(__UNIX_JACK__) + +// JACK is a low-latency audio server, originally written for the +// GNU/Linux operating system and now also ported to OS-X. It can +// connect a number of different applications to an audio device, as +// well as allowing them to share audio between themselves. +// +// When using JACK with RtAudio, "devices" refer to JACK clients that +// have ports connected to the server. The JACK server is typically +// started in a terminal as follows: +// +// .jackd -d alsa -d hw:0 +// +// or through an interface program such as qjackctl. Many of the +// parameters normally set for a stream are fixed by the JACK server +// and can be specified when the JACK server is started. In +// particular, +// +// .jackd -d alsa -d hw:0 -r 44100 -p 512 -n 4 +// +// specifies a sample rate of 44100 Hz, a buffer size of 512 sample +// frames, and number of buffers = 4. Once the server is running, it +// is not possible to override these values. If the values are not +// specified in the command-line, the JACK server uses default values. +// +// The JACK server does not have to be running when an instance of +// RtApiJack is created, though the function getDeviceCount() will +// report 0 devices found until JACK has been started. When no +// devices are available (i.e., the JACK server is not running), a +// stream cannot be opened. + +#include <jack/jack.h> +#include <unistd.h> +#include <cstdio> + +// A structure to hold various information related to the Jack API +// implementation. +struct JackHandle { + jack_client_t *client; + jack_port_t **ports[2]; + std::string deviceName[2]; + bool xrun[2]; + pthread_cond_t condition; + int drainCounter; // Tracks callback counts when draining + bool internalDrain; // Indicates if stop is initiated from callback or not. + + JackHandle() + :client(0), drainCounter(0), internalDrain(false) { ports[0] = 0; ports[1] = 0; xrun[0] = false; xrun[1] = false; } +}; + +#if !defined(__RTAUDIO_DEBUG__) +static void jackSilentError( const char * ) {}; +#endif + +RtApiJack :: RtApiJack() + :shouldAutoconnect_(true) { + // Nothing to do here. +#if !defined(__RTAUDIO_DEBUG__) + // Turn off Jack's internal error reporting. + jack_set_error_function( &jackSilentError ); +#endif +} + +RtApiJack :: ~RtApiJack() +{ + if ( stream_.state != STREAM_CLOSED ) closeStream(); +} + +unsigned int RtApiJack :: getDeviceCount( void ) +{ + // See if we can become a jack client. + jack_options_t options = (jack_options_t) ( JackNoStartServer ); //JackNullOption; + jack_status_t *status = NULL; + jack_client_t *client = jack_client_open( "RtApiJackCount", options, status ); + if ( client == 0 ) return 0; + + const char **ports; + std::string port, previousPort; + unsigned int nChannels = 0, nDevices = 0; + ports = jack_get_ports( client, NULL, JACK_DEFAULT_AUDIO_TYPE, 0 ); + if ( ports ) { + // Parse the port names up to the first colon (:). + size_t iColon = 0; + do { + port = (char *) ports[ nChannels ]; + iColon = port.find(":"); + if ( iColon != std::string::npos ) { + port = port.substr( 0, iColon + 1 ); + if ( port != previousPort ) { + nDevices++; + previousPort = port; + } + } + } while ( ports[++nChannels] ); + free( ports ); + } + + jack_client_close( client ); + return nDevices; +} + +RtAudio::DeviceInfo RtApiJack :: getDeviceInfo( unsigned int device ) +{ + RtAudio::DeviceInfo info; + info.probed = false; + + jack_options_t options = (jack_options_t) ( JackNoStartServer ); //JackNullOption + jack_status_t *status = NULL; + jack_client_t *client = jack_client_open( "RtApiJackInfo", options, status ); + if ( client == 0 ) { + errorText_ = "RtApiJack::getDeviceInfo: Jack server not found or connection error!"; + error( RtAudioError::WARNING ); + return info; + } + + const char **ports; + std::string port, previousPort; + unsigned int nPorts = 0, nDevices = 0; + ports = jack_get_ports( client, NULL, JACK_DEFAULT_AUDIO_TYPE, 0 ); + if ( ports ) { + // Parse the port names up to the first colon (:). + size_t iColon = 0; + do { + port = (char *) ports[ nPorts ]; + iColon = port.find(":"); + if ( iColon != std::string::npos ) { + port = port.substr( 0, iColon ); + if ( port != previousPort ) { + if ( nDevices == device ) info.name = port; + nDevices++; + previousPort = port; + } + } + } while ( ports[++nPorts] ); + free( ports ); + } + + if ( device >= nDevices ) { + jack_client_close( client ); + errorText_ = "RtApiJack::getDeviceInfo: device ID is invalid!"; + error( RtAudioError::INVALID_USE ); + return info; + } + + // Get the current jack server sample rate. + info.sampleRates.clear(); + + info.preferredSampleRate = jack_get_sample_rate( client ); + info.sampleRates.push_back( info.preferredSampleRate ); + + // Count the available ports containing the client name as device + // channels. Jack "input ports" equal RtAudio output channels. + unsigned int nChannels = 0; + ports = jack_get_ports( client, info.name.c_str(), JACK_DEFAULT_AUDIO_TYPE, JackPortIsInput ); + if ( ports ) { + while ( ports[ nChannels ] ) nChannels++; + free( ports ); + info.outputChannels = nChannels; + } + + // Jack "output ports" equal RtAudio input channels. + nChannels = 0; + ports = jack_get_ports( client, info.name.c_str(), JACK_DEFAULT_AUDIO_TYPE, JackPortIsOutput ); + if ( ports ) { + while ( ports[ nChannels ] ) nChannels++; + free( ports ); + info.inputChannels = nChannels; + } + + if ( info.outputChannels == 0 && info.inputChannels == 0 ) { + jack_client_close(client); + errorText_ = "RtApiJack::getDeviceInfo: error determining Jack input/output channels!"; + error( RtAudioError::WARNING ); + return info; + } + + // If device opens for both playback and capture, we determine the channels. + if ( info.outputChannels > 0 && info.inputChannels > 0 ) + info.duplexChannels = (info.outputChannels > info.inputChannels) ? info.inputChannels : info.outputChannels; + + // Jack always uses 32-bit floats. + info.nativeFormats = RTAUDIO_FLOAT32; + + // Jack doesn't provide default devices so we'll use the first available one. + if ( device == 0 && info.outputChannels > 0 ) + info.isDefaultOutput = true; + if ( device == 0 && info.inputChannels > 0 ) + info.isDefaultInput = true; + + jack_client_close(client); + info.probed = true; + return info; +} + +static int jackCallbackHandler( jack_nframes_t nframes, void *infoPointer ) +{ + CallbackInfo *info = (CallbackInfo *) infoPointer; + + RtApiJack *object = (RtApiJack *) info->object; + if ( object->callbackEvent( (unsigned long) nframes ) == false ) return 1; + + return 0; +} + +// This function will be called by a spawned thread when the Jack +// server signals that it is shutting down. It is necessary to handle +// it this way because the jackShutdown() function must return before +// the jack_deactivate() function (in closeStream()) will return. +static void *jackCloseStream( void *ptr ) +{ + CallbackInfo *info = (CallbackInfo *) ptr; + RtApiJack *object = (RtApiJack *) info->object; + + object->closeStream(); + + pthread_exit( NULL ); +} +static void jackShutdown( void *infoPointer ) +{ + CallbackInfo *info = (CallbackInfo *) infoPointer; + RtApiJack *object = (RtApiJack *) info->object; + + // Check current stream state. If stopped, then we'll assume this + // was called as a result of a call to RtApiJack::stopStream (the + // deactivation of a client handle causes this function to be called). + // If not, we'll assume the Jack server is shutting down or some + // other problem occurred and we should close the stream. + if ( object->isStreamRunning() == false ) return; + + ThreadHandle threadId; + pthread_create( &threadId, NULL, jackCloseStream, info ); + std::cerr << "\nRtApiJack: the Jack server is shutting down this client ... stream stopped and closed!!\n" << std::endl; +} + +static int jackXrun( void *infoPointer ) +{ + JackHandle *handle = *((JackHandle **) infoPointer); + + if ( handle->ports[0] ) handle->xrun[0] = true; + if ( handle->ports[1] ) handle->xrun[1] = true; + + return 0; +} + +bool RtApiJack :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels, + unsigned int firstChannel, unsigned int sampleRate, + RtAudioFormat format, unsigned int *bufferSize, + RtAudio::StreamOptions *options ) +{ + JackHandle *handle = (JackHandle *) stream_.apiHandle; + + // Look for jack server and try to become a client (only do once per stream). + jack_client_t *client = 0; + if ( mode == OUTPUT || ( mode == INPUT && stream_.mode != OUTPUT ) ) { + jack_options_t jackoptions = (jack_options_t) ( JackNoStartServer ); //JackNullOption; + jack_status_t *status = NULL; + if ( options && !options->streamName.empty() ) + client = jack_client_open( options->streamName.c_str(), jackoptions, status ); + else + client = jack_client_open( "RtApiJack", jackoptions, status ); + if ( client == 0 ) { + errorText_ = "RtApiJack::probeDeviceOpen: Jack server not found or connection error!"; + error( RtAudioError::WARNING ); + return FAILURE; + } + } + else { + // The handle must have been created on an earlier pass. + client = handle->client; + } + + const char **ports; + std::string port, previousPort, deviceName; + unsigned int nPorts = 0, nDevices = 0; + ports = jack_get_ports( client, NULL, JACK_DEFAULT_AUDIO_TYPE, 0 ); + if ( ports ) { + // Parse the port names up to the first colon (:). + size_t iColon = 0; + do { + port = (char *) ports[ nPorts ]; + iColon = port.find(":"); + if ( iColon != std::string::npos ) { + port = port.substr( 0, iColon ); + if ( port != previousPort ) { + if ( nDevices == device ) deviceName = port; + nDevices++; + previousPort = port; + } + } + } while ( ports[++nPorts] ); + free( ports ); + } + + if ( device >= nDevices ) { + errorText_ = "RtApiJack::probeDeviceOpen: device ID is invalid!"; + return FAILURE; + } + + unsigned long flag = JackPortIsInput; + if ( mode == INPUT ) flag = JackPortIsOutput; + + if ( ! (options && (options->flags & RTAUDIO_JACK_DONT_CONNECT)) ) { + // Count the available ports containing the client name as device + // channels. Jack "input ports" equal RtAudio output channels. + unsigned int nChannels = 0; + ports = jack_get_ports( client, deviceName.c_str(), JACK_DEFAULT_AUDIO_TYPE, flag ); + if ( ports ) { + while ( ports[ nChannels ] ) nChannels++; + free( ports ); + } + // Compare the jack ports for specified client to the requested number of channels. + if ( nChannels < (channels + firstChannel) ) { + errorStream_ << "RtApiJack::probeDeviceOpen: requested number of channels (" << channels << ") + offset (" << firstChannel << ") not found for specified device (" << device << ":" << deviceName << ")."; + errorText_ = errorStream_.str(); + return FAILURE; + } + } + + // Check the jack server sample rate. + unsigned int jackRate = jack_get_sample_rate( client ); + if ( sampleRate != jackRate ) { + jack_client_close( client ); + errorStream_ << "RtApiJack::probeDeviceOpen: the requested sample rate (" << sampleRate << ") is different than the JACK server rate (" << jackRate << ")."; + errorText_ = errorStream_.str(); + return FAILURE; + } + stream_.sampleRate = jackRate; + + // Get the latency of the JACK port. + ports = jack_get_ports( client, deviceName.c_str(), JACK_DEFAULT_AUDIO_TYPE, flag ); + if ( ports[ firstChannel ] ) { + // Added by Ge Wang + jack_latency_callback_mode_t cbmode = (mode == INPUT ? JackCaptureLatency : JackPlaybackLatency); + // the range (usually the min and max are equal) + jack_latency_range_t latrange; latrange.min = latrange.max = 0; + // get the latency range + jack_port_get_latency_range( jack_port_by_name( client, ports[firstChannel] ), cbmode, &latrange ); + // be optimistic, use the min! + stream_.latency[mode] = latrange.min; + //stream_.latency[mode] = jack_port_get_latency( jack_port_by_name( client, ports[ firstChannel ] ) ); + } + free( ports ); + + // The jack server always uses 32-bit floating-point data. + stream_.deviceFormat[mode] = RTAUDIO_FLOAT32; + stream_.userFormat = format; + + if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) stream_.userInterleaved = false; + else stream_.userInterleaved = true; + + // Jack always uses non-interleaved buffers. + stream_.deviceInterleaved[mode] = false; + + // Jack always provides host byte-ordered data. + stream_.doByteSwap[mode] = false; + + // Get the buffer size. The buffer size and number of buffers + // (periods) is set when the jack server is started. + stream_.bufferSize = (int) jack_get_buffer_size( client ); + *bufferSize = stream_.bufferSize; + + stream_.nDeviceChannels[mode] = channels; + stream_.nUserChannels[mode] = channels; + + // Set flags for buffer conversion. + stream_.doConvertBuffer[mode] = false; + if ( stream_.userFormat != stream_.deviceFormat[mode] ) + stream_.doConvertBuffer[mode] = true; + if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] && + stream_.nUserChannels[mode] > 1 ) + stream_.doConvertBuffer[mode] = true; + + // Allocate our JackHandle structure for the stream. + if ( handle == 0 ) { + try { + handle = new JackHandle; + } + catch ( std::bad_alloc& ) { + errorText_ = "RtApiJack::probeDeviceOpen: error allocating JackHandle memory."; + goto error; + } + + if ( pthread_cond_init(&handle->condition, NULL) ) { + errorText_ = "RtApiJack::probeDeviceOpen: error initializing pthread condition variable."; + goto error; + } + stream_.apiHandle = (void *) handle; + handle->client = client; + } + handle->deviceName[mode] = deviceName; + + // Allocate necessary internal buffers. + unsigned long bufferBytes; + bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat ); + stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 ); + if ( stream_.userBuffer[mode] == NULL ) { + errorText_ = "RtApiJack::probeDeviceOpen: error allocating user buffer memory."; + goto error; + } + + if ( stream_.doConvertBuffer[mode] ) { + + bool makeBuffer = true; + if ( mode == OUTPUT ) + bufferBytes = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] ); + else { // mode == INPUT + bufferBytes = stream_.nDeviceChannels[1] * formatBytes( stream_.deviceFormat[1] ); + if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) { + unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes(stream_.deviceFormat[0]); + if ( bufferBytes < bytesOut ) makeBuffer = false; + } + } + + if ( makeBuffer ) { + bufferBytes *= *bufferSize; + if ( stream_.deviceBuffer ) free( stream_.deviceBuffer ); + stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 ); + if ( stream_.deviceBuffer == NULL ) { + errorText_ = "RtApiJack::probeDeviceOpen: error allocating device buffer memory."; + goto error; + } + } + } + + // Allocate memory for the Jack ports (channels) identifiers. + handle->ports[mode] = (jack_port_t **) malloc ( sizeof (jack_port_t *) * channels ); + if ( handle->ports[mode] == NULL ) { + errorText_ = "RtApiJack::probeDeviceOpen: error allocating port memory."; + goto error; + } + + stream_.device[mode] = device; + stream_.channelOffset[mode] = firstChannel; + stream_.state = STREAM_STOPPED; + stream_.callbackInfo.object = (void *) this; + + if ( stream_.mode == OUTPUT && mode == INPUT ) + // We had already set up the stream for output. + stream_.mode = DUPLEX; + else { + stream_.mode = mode; + jack_set_process_callback( handle->client, jackCallbackHandler, (void *) &stream_.callbackInfo ); + jack_set_xrun_callback( handle->client, jackXrun, (void *) &stream_.apiHandle ); + jack_on_shutdown( handle->client, jackShutdown, (void *) &stream_.callbackInfo ); + } + + // Register our ports. + char label[64]; + if ( mode == OUTPUT ) { + for ( unsigned int i=0; i<stream_.nUserChannels[0]; i++ ) { + snprintf( label, 64, "outport %d", i ); + handle->ports[0][i] = jack_port_register( handle->client, (const char *)label, + JACK_DEFAULT_AUDIO_TYPE, JackPortIsOutput, 0 ); + } + } + else { + for ( unsigned int i=0; i<stream_.nUserChannels[1]; i++ ) { + snprintf( label, 64, "inport %d", i ); + handle->ports[1][i] = jack_port_register( handle->client, (const char *)label, + JACK_DEFAULT_AUDIO_TYPE, JackPortIsInput, 0 ); + } + } + + // Setup the buffer conversion information structure. We don't use + // buffers to do channel offsets, so we override that parameter + // here. + if ( stream_.doConvertBuffer[mode] ) setConvertInfo( mode, 0 ); + + if ( options && options->flags & RTAUDIO_JACK_DONT_CONNECT ) shouldAutoconnect_ = false; + + return SUCCESS; + + error: + if ( handle ) { + pthread_cond_destroy( &handle->condition ); + jack_client_close( handle->client ); + + if ( handle->ports[0] ) free( handle->ports[0] ); + if ( handle->ports[1] ) free( handle->ports[1] ); + + delete handle; + stream_.apiHandle = 0; + } + + for ( int i=0; i<2; i++ ) { + if ( stream_.userBuffer[i] ) { + free( stream_.userBuffer[i] ); + stream_.userBuffer[i] = 0; + } + } + + if ( stream_.deviceBuffer ) { + free( stream_.deviceBuffer ); + stream_.deviceBuffer = 0; + } + + return FAILURE; +} + +void RtApiJack :: closeStream( void ) +{ + if ( stream_.state == STREAM_CLOSED ) { + errorText_ = "RtApiJack::closeStream(): no open stream to close!"; + error( RtAudioError::WARNING ); + return; + } + + JackHandle *handle = (JackHandle *) stream_.apiHandle; + if ( handle ) { + + if ( stream_.state == STREAM_RUNNING ) + jack_deactivate( handle->client ); + + jack_client_close( handle->client ); + } + + if ( handle ) { + if ( handle->ports[0] ) free( handle->ports[0] ); + if ( handle->ports[1] ) free( handle->ports[1] ); + pthread_cond_destroy( &handle->condition ); + delete handle; + stream_.apiHandle = 0; + } + + for ( int i=0; i<2; i++ ) { + if ( stream_.userBuffer[i] ) { + free( stream_.userBuffer[i] ); + stream_.userBuffer[i] = 0; + } + } + + if ( stream_.deviceBuffer ) { + free( stream_.deviceBuffer ); + stream_.deviceBuffer = 0; + } + + stream_.mode = UNINITIALIZED; + stream_.state = STREAM_CLOSED; +} + +void RtApiJack :: startStream( void ) +{ + verifyStream(); + if ( stream_.state == STREAM_RUNNING ) { + errorText_ = "RtApiJack::startStream(): the stream is already running!"; + error( RtAudioError::WARNING ); + return; + } + + #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) ) { + result = 1; + ports = jack_get_ports( handle->client, handle->deviceName[0].c_str(), JACK_DEFAULT_AUDIO_TYPE, JackPortIsInput); + if ( ports == NULL) { + errorText_ = "RtApiJack::startStream(): error determining available JACK input ports!"; + goto unlock; + } + + // Now make the port connections. Since RtAudio wasn't designed to + // allow the user to select particular channels of a device, we'll + // just open the first "nChannels" ports with offset. + for ( unsigned int i=0; i<stream_.nUserChannels[0]; i++ ) { + result = 1; + if ( ports[ stream_.channelOffset[0] + i ] ) + result = jack_connect( handle->client, jack_port_name( handle->ports[0][i] ), ports[ stream_.channelOffset[0] + i ] ); + if ( result ) { + free( ports ); + errorText_ = "RtApiJack::startStream(): error connecting output ports!"; + goto unlock; + } + } + free(ports); + } + + if ( shouldAutoconnect_ && (stream_.mode == INPUT || stream_.mode == DUPLEX) ) { + result = 1; + ports = jack_get_ports( handle->client, handle->deviceName[1].c_str(), JACK_DEFAULT_AUDIO_TYPE, JackPortIsOutput ); + if ( ports == NULL) { + errorText_ = "RtApiJack::startStream(): error determining available JACK output ports!"; + goto unlock; + } + + // Now make the port connections. See note above. + for ( unsigned int i=0; i<stream_.nUserChannels[1]; i++ ) { + result = 1; + if ( ports[ stream_.channelOffset[1] + i ] ) + result = jack_connect( handle->client, ports[ stream_.channelOffset[1] + i ], jack_port_name( handle->ports[1][i] ) ); + if ( result ) { + free( ports ); + errorText_ = "RtApiJack::startStream(): error connecting input ports!"; + goto unlock; + } + } + free(ports); + } + + handle->drainCounter = 0; + handle->internalDrain = false; + stream_.state = STREAM_RUNNING; + + unlock: + if ( result == 0 ) return; + error( RtAudioError::SYSTEM_ERROR ); +} + +void RtApiJack :: stopStream( void ) +{ + verifyStream(); + if ( stream_.state == STREAM_STOPPED ) { + errorText_ = "RtApiJack::stopStream(): the stream is already stopped!"; + error( RtAudioError::WARNING ); + return; + } + + JackHandle *handle = (JackHandle *) stream_.apiHandle; + if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { + + if ( handle->drainCounter == 0 ) { + handle->drainCounter = 2; + pthread_cond_wait( &handle->condition, &stream_.mutex ); // block until signaled + } + } + + jack_deactivate( handle->client ); + stream_.state = STREAM_STOPPED; +} + +void RtApiJack :: abortStream( void ) +{ + verifyStream(); + if ( stream_.state == STREAM_STOPPED ) { + errorText_ = "RtApiJack::abortStream(): the stream is already stopped!"; + error( RtAudioError::WARNING ); + return; + } + + JackHandle *handle = (JackHandle *) stream_.apiHandle; + handle->drainCounter = 2; + + stopStream(); +} + +// This function will be called by a spawned thread when the user +// callback function signals that the stream should be stopped or +// aborted. It is necessary to handle it this way because the +// callbackEvent() function must return before the jack_deactivate() +// function will return. +static void *jackStopStream( void *ptr ) +{ + CallbackInfo *info = (CallbackInfo *) ptr; + RtApiJack *object = (RtApiJack *) info->object; + + object->stopStream(); + pthread_exit( NULL ); +} + +bool RtApiJack :: callbackEvent( unsigned long nframes ) +{ + if ( stream_.state == STREAM_STOPPED || stream_.state == STREAM_STOPPING ) return SUCCESS; + if ( stream_.state == STREAM_CLOSED ) { + errorText_ = "RtApiCore::callbackEvent(): the stream is closed ... this shouldn't happen!"; + error( RtAudioError::WARNING ); + return FAILURE; + } + if ( stream_.bufferSize != nframes ) { + errorText_ = "RtApiCore::callbackEvent(): the JACK buffer size has changed ... cannot process!"; + error( RtAudioError::WARNING ); + return FAILURE; + } + + CallbackInfo *info = (CallbackInfo *) &stream_.callbackInfo; + JackHandle *handle = (JackHandle *) stream_.apiHandle; + + // Check if we were draining the stream and signal is finished. + if ( handle->drainCounter > 3 ) { + ThreadHandle threadId; + + stream_.state = STREAM_STOPPING; + if ( handle->internalDrain == true ) + pthread_create( &threadId, NULL, jackStopStream, info ); + else + pthread_cond_signal( &handle->condition ); + return SUCCESS; + } + + // Invoke user callback first, to get fresh output data. + if ( handle->drainCounter == 0 ) { + RtAudioCallback callback = (RtAudioCallback) info->callback; + double streamTime = getStreamTime(); + RtAudioStreamStatus status = 0; + if ( stream_.mode != INPUT && handle->xrun[0] == true ) { + status |= RTAUDIO_OUTPUT_UNDERFLOW; + handle->xrun[0] = false; + } + if ( stream_.mode != OUTPUT && handle->xrun[1] == true ) { + status |= RTAUDIO_INPUT_OVERFLOW; + handle->xrun[1] = false; + } + int cbReturnValue = callback( stream_.userBuffer[0], stream_.userBuffer[1], + stream_.bufferSize, streamTime, status, info->userData ); + if ( cbReturnValue == 2 ) { + stream_.state = STREAM_STOPPING; + handle->drainCounter = 2; + ThreadHandle id; + pthread_create( &id, NULL, jackStopStream, info ); + return SUCCESS; + } + else if ( cbReturnValue == 1 ) { + handle->drainCounter = 1; + handle->internalDrain = true; + } + } + + jack_default_audio_sample_t *jackbuffer; + unsigned long bufferBytes = nframes * sizeof( jack_default_audio_sample_t ); + if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { + + if ( handle->drainCounter > 1 ) { // write zeros to the output stream + + for ( unsigned int i=0; i<stream_.nDeviceChannels[0]; i++ ) { + jackbuffer = (jack_default_audio_sample_t *) jack_port_get_buffer( handle->ports[0][i], (jack_nframes_t) nframes ); + memset( jackbuffer, 0, bufferBytes ); + } + + } + else if ( stream_.doConvertBuffer[0] ) { + + convertBuffer( stream_.deviceBuffer, stream_.userBuffer[0], stream_.convertInfo[0] ); + + for ( unsigned int i=0; i<stream_.nDeviceChannels[0]; i++ ) { + jackbuffer = (jack_default_audio_sample_t *) jack_port_get_buffer( handle->ports[0][i], (jack_nframes_t) nframes ); + memcpy( jackbuffer, &stream_.deviceBuffer[i*bufferBytes], bufferBytes ); + } + } + else { // no buffer conversion + for ( unsigned int i=0; i<stream_.nUserChannels[0]; i++ ) { + jackbuffer = (jack_default_audio_sample_t *) jack_port_get_buffer( handle->ports[0][i], (jack_nframes_t) nframes ); + memcpy( jackbuffer, &stream_.userBuffer[0][i*bufferBytes], bufferBytes ); + } + } + } + + // Don't bother draining input + if ( handle->drainCounter ) { + handle->drainCounter++; + goto unlock; + } + + if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) { + + if ( stream_.doConvertBuffer[1] ) { + for ( unsigned int i=0; i<stream_.nDeviceChannels[1]; i++ ) { + jackbuffer = (jack_default_audio_sample_t *) jack_port_get_buffer( handle->ports[1][i], (jack_nframes_t) nframes ); + memcpy( &stream_.deviceBuffer[i*bufferBytes], jackbuffer, bufferBytes ); + } + convertBuffer( stream_.userBuffer[1], stream_.deviceBuffer, stream_.convertInfo[1] ); + } + else { // no buffer conversion + for ( unsigned int i=0; i<stream_.nUserChannels[1]; i++ ) { + jackbuffer = (jack_default_audio_sample_t *) jack_port_get_buffer( handle->ports[1][i], (jack_nframes_t) nframes ); + memcpy( &stream_.userBuffer[1][i*bufferBytes], jackbuffer, bufferBytes ); + } + } + } + + unlock: + RtApi::tickStreamTime(); + return SUCCESS; +} + //******************** End of __UNIX_JACK__ *********************// +#endif + +#if defined(__WINDOWS_ASIO__) // ASIO API on Windows + +// The ASIO API is designed around a callback scheme, so this +// implementation is similar to that used for OS-X CoreAudio and Linux +// Jack. The primary constraint with ASIO is that it only allows +// access to a single driver at a time. Thus, it is not possible to +// have more than one simultaneous RtAudio stream. +// +// This implementation also requires a number of external ASIO files +// and a few global variables. The ASIO callback scheme does not +// allow for the passing of user data, so we must create a global +// pointer to our callbackInfo structure. +// +// On unix systems, we make use of a pthread condition variable. +// Since there is no equivalent in Windows, I hacked something based +// on information found in +// http://www.cs.wustl.edu/~schmidt/win32-cv-1.html. + +#include "asiosys.h" +#include "asio.h" +#include "iasiothiscallresolver.h" +#include "asiodrivers.h" +#include <cmath> + +static AsioDrivers drivers; +static ASIOCallbacks asioCallbacks; +static ASIODriverInfo driverInfo; +static CallbackInfo *asioCallbackInfo; +static bool asioXRun; + +struct AsioHandle { + int drainCounter; // Tracks callback counts when draining + bool internalDrain; // Indicates if stop is initiated from callback or not. + ASIOBufferInfo *bufferInfos; + HANDLE condition; + + AsioHandle() + :drainCounter(0), internalDrain(false), bufferInfos(0) {} +}; + +// Function declarations (definitions at end of section) +static const char* getAsioErrorString( ASIOError result ); +static void sampleRateChanged( ASIOSampleRate sRate ); +static long asioMessages( long selector, long value, void* message, double* opt ); + +RtApiAsio :: RtApiAsio() +{ + // ASIO cannot run on a multi-threaded appartment. You can call + // CoInitialize beforehand, but it must be for appartment threading + // (in which case, CoInitilialize will return S_FALSE here). + coInitialized_ = false; + HRESULT hr = CoInitialize( NULL ); + if ( FAILED(hr) ) { + errorText_ = "RtApiAsio::ASIO requires a single-threaded appartment. Call CoInitializeEx(0,COINIT_APARTMENTTHREADED)"; + error( RtAudioError::WARNING ); + } + coInitialized_ = true; + + drivers.removeCurrentDriver(); + driverInfo.asioVersion = 2; + + // See note in DirectSound implementation about GetDesktopWindow(). + driverInfo.sysRef = GetForegroundWindow(); +} + +RtApiAsio :: ~RtApiAsio() +{ + if ( stream_.state != STREAM_CLOSED ) closeStream(); + if ( coInitialized_ ) CoUninitialize(); +} + +unsigned int RtApiAsio :: getDeviceCount( void ) +{ + return (unsigned int) drivers.asioGetNumDev(); +} + +RtAudio::DeviceInfo RtApiAsio :: getDeviceInfo( unsigned int device ) +{ + RtAudio::DeviceInfo info; + info.probed = false; + + // Get device ID + unsigned int nDevices = getDeviceCount(); + if ( nDevices == 0 ) { + errorText_ = "RtApiAsio::getDeviceInfo: no devices found!"; + error( RtAudioError::INVALID_USE ); + return info; + } + + if ( device >= nDevices ) { + errorText_ = "RtApiAsio::getDeviceInfo: device ID is invalid!"; + error( RtAudioError::INVALID_USE ); + return info; + } + + // If a stream is already open, we cannot probe other devices. Thus, use the saved results. + if ( stream_.state != STREAM_CLOSED ) { + if ( device >= devices_.size() ) { + errorText_ = "RtApiAsio::getDeviceInfo: device ID was not present before stream was opened."; + error( RtAudioError::WARNING ); + return info; + } + return devices_[ device ]; + } + + char driverName[32]; + ASIOError result = drivers.asioGetDriverName( (int) device, driverName, 32 ); + if ( result != ASE_OK ) { + errorStream_ << "RtApiAsio::getDeviceInfo: unable to get driver name (" << getAsioErrorString( result ) << ")."; + errorText_ = errorStream_.str(); + error( RtAudioError::WARNING ); + return info; + } + + info.name = driverName; + + if ( !drivers.loadDriver( driverName ) ) { + errorStream_ << "RtApiAsio::getDeviceInfo: unable to load driver (" << driverName << ")."; + errorText_ = errorStream_.str(); + error( RtAudioError::WARNING ); + return info; + } + + result = ASIOInit( &driverInfo ); + if ( result != ASE_OK ) { + errorStream_ << "RtApiAsio::getDeviceInfo: error (" << getAsioErrorString( result ) << ") initializing driver (" << driverName << ")."; + errorText_ = errorStream_.str(); + error( RtAudioError::WARNING ); + return info; + } + + // Determine the device channel information. + long inputChannels, outputChannels; + result = ASIOGetChannels( &inputChannels, &outputChannels ); + if ( result != ASE_OK ) { + drivers.removeCurrentDriver(); + errorStream_ << "RtApiAsio::getDeviceInfo: error (" << getAsioErrorString( result ) << ") getting channel count (" << driverName << ")."; + errorText_ = errorStream_.str(); + error( RtAudioError::WARNING ); + return info; + } + + info.outputChannels = outputChannels; + info.inputChannels = inputChannels; + if ( info.outputChannels > 0 && info.inputChannels > 0 ) + info.duplexChannels = (info.outputChannels > info.inputChannels) ? info.inputChannels : info.outputChannels; + + // Determine the supported sample rates. + info.sampleRates.clear(); + for ( unsigned int i=0; i<MAX_SAMPLE_RATES; i++ ) { + result = ASIOCanSampleRate( (ASIOSampleRate) SAMPLE_RATES[i] ); + if ( result == ASE_OK ) { + info.sampleRates.push_back( SAMPLE_RATES[i] ); + + if ( !info.preferredSampleRate || ( SAMPLE_RATES[i] <= 48000 && SAMPLE_RATES[i] > info.preferredSampleRate ) ) + info.preferredSampleRate = SAMPLE_RATES[i]; + } + } + + // Determine supported data types ... just check first channel and assume rest are the same. + ASIOChannelInfo channelInfo; + channelInfo.channel = 0; + channelInfo.isInput = true; + if ( info.inputChannels <= 0 ) channelInfo.isInput = false; + result = ASIOGetChannelInfo( &channelInfo ); + if ( result != ASE_OK ) { + drivers.removeCurrentDriver(); + errorStream_ << "RtApiAsio::getDeviceInfo: error (" << getAsioErrorString( result ) << ") getting driver channel info (" << driverName << ")."; + errorText_ = errorStream_.str(); + error( RtAudioError::WARNING ); + return info; + } + + info.nativeFormats = 0; + if ( channelInfo.type == ASIOSTInt16MSB || channelInfo.type == ASIOSTInt16LSB ) + info.nativeFormats |= RTAUDIO_SINT16; + else if ( channelInfo.type == ASIOSTInt32MSB || channelInfo.type == ASIOSTInt32LSB ) + info.nativeFormats |= RTAUDIO_SINT32; + else if ( channelInfo.type == ASIOSTFloat32MSB || channelInfo.type == ASIOSTFloat32LSB ) + info.nativeFormats |= RTAUDIO_FLOAT32; + else if ( channelInfo.type == ASIOSTFloat64MSB || channelInfo.type == ASIOSTFloat64LSB ) + info.nativeFormats |= RTAUDIO_FLOAT64; + else if ( channelInfo.type == ASIOSTInt24MSB || channelInfo.type == ASIOSTInt24LSB ) + info.nativeFormats |= RTAUDIO_SINT24; + + if ( info.outputChannels > 0 ) + if ( getDefaultOutputDevice() == device ) info.isDefaultOutput = true; + if ( info.inputChannels > 0 ) + if ( getDefaultInputDevice() == device ) info.isDefaultInput = true; + + info.probed = true; + drivers.removeCurrentDriver(); + return info; +} + +static void bufferSwitch( long index, ASIOBool /*processNow*/ ) +{ + RtApiAsio *object = (RtApiAsio *) asioCallbackInfo->object; + object->callbackEvent( index ); +} + +void RtApiAsio :: saveDeviceInfo( void ) +{ + devices_.clear(); + + unsigned int nDevices = getDeviceCount(); + devices_.resize( nDevices ); + for ( unsigned int i=0; i<nDevices; i++ ) + devices_[i] = getDeviceInfo( i ); +} + +bool RtApiAsio :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels, + unsigned int firstChannel, unsigned int sampleRate, + RtAudioFormat format, unsigned int *bufferSize, + RtAudio::StreamOptions *options ) +{//////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////// + + bool isDuplexInput = mode == INPUT && stream_.mode == OUTPUT; + + // For ASIO, a duplex stream MUST use the same driver. + if ( isDuplexInput && stream_.device[0] != device ) { + errorText_ = "RtApiAsio::probeDeviceOpen: an ASIO duplex stream must use the same device for input and output!"; + return FAILURE; + } + + char driverName[32]; + ASIOError result = drivers.asioGetDriverName( (int) device, driverName, 32 ); + if ( result != ASE_OK ) { + errorStream_ << "RtApiAsio::probeDeviceOpen: unable to get driver name (" << getAsioErrorString( result ) << ")."; + errorText_ = errorStream_.str(); + return FAILURE; + } + + // Only load the driver once for duplex stream. + if ( !isDuplexInput ) { + // The getDeviceInfo() function will not work when a stream is open + // because ASIO does not allow multiple devices to run at the same + // time. Thus, we'll probe the system before opening a stream and + // save the results for use by getDeviceInfo(). + this->saveDeviceInfo(); + + if ( !drivers.loadDriver( driverName ) ) { + errorStream_ << "RtApiAsio::probeDeviceOpen: unable to load driver (" << driverName << ")."; + errorText_ = errorStream_.str(); + return FAILURE; + } + + result = ASIOInit( &driverInfo ); + if ( result != ASE_OK ) { + errorStream_ << "RtApiAsio::probeDeviceOpen: error (" << getAsioErrorString( result ) << ") initializing driver (" << driverName << ")."; + errorText_ = errorStream_.str(); + return FAILURE; + } + } + + // keep them before any "goto error", they are used for error cleanup + goto device boundary checks + bool buffersAllocated = false; + AsioHandle *handle = (AsioHandle *) stream_.apiHandle; + unsigned int nChannels; + + + // Check the device channel count. + long inputChannels, outputChannels; + result = ASIOGetChannels( &inputChannels, &outputChannels ); + if ( result != ASE_OK ) { + errorStream_ << "RtApiAsio::probeDeviceOpen: error (" << getAsioErrorString( result ) << ") getting channel count (" << driverName << ")."; + errorText_ = errorStream_.str(); + goto error; + } + + if ( ( mode == OUTPUT && (channels+firstChannel) > (unsigned int) outputChannels) || + ( mode == INPUT && (channels+firstChannel) > (unsigned int) inputChannels) ) { + errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") does not support requested channel count (" << channels << ") + offset (" << firstChannel << ")."; + errorText_ = errorStream_.str(); + goto error; + } + stream_.nDeviceChannels[mode] = channels; + stream_.nUserChannels[mode] = channels; + stream_.channelOffset[mode] = firstChannel; + + // Verify the sample rate is supported. + result = ASIOCanSampleRate( (ASIOSampleRate) sampleRate ); + if ( result != ASE_OK ) { + errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") does not support requested sample rate (" << sampleRate << ")."; + errorText_ = errorStream_.str(); + goto error; + } + + // Get the current sample rate + ASIOSampleRate currentRate; + result = ASIOGetSampleRate( ¤tRate ); + if ( result != ASE_OK ) { + errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error getting sample rate."; + errorText_ = errorStream_.str(); + goto error; + } + + // Set the sample rate only if necessary + if ( currentRate != sampleRate ) { + result = ASIOSetSampleRate( (ASIOSampleRate) sampleRate ); + if ( result != ASE_OK ) { + errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error setting sample rate (" << sampleRate << ")."; + errorText_ = errorStream_.str(); + goto error; + } + } + + // Determine the driver data type. + ASIOChannelInfo channelInfo; + channelInfo.channel = 0; + if ( mode == OUTPUT ) channelInfo.isInput = false; + else channelInfo.isInput = true; + result = ASIOGetChannelInfo( &channelInfo ); + if ( result != ASE_OK ) { + errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error (" << getAsioErrorString( result ) << ") getting data format."; + errorText_ = errorStream_.str(); + goto error; + } + + // Assuming WINDOWS host is always little-endian. + stream_.doByteSwap[mode] = false; + stream_.userFormat = format; + stream_.deviceFormat[mode] = 0; + if ( channelInfo.type == ASIOSTInt16MSB || channelInfo.type == ASIOSTInt16LSB ) { + stream_.deviceFormat[mode] = RTAUDIO_SINT16; + if ( channelInfo.type == ASIOSTInt16MSB ) stream_.doByteSwap[mode] = true; + } + else if ( channelInfo.type == ASIOSTInt32MSB || channelInfo.type == ASIOSTInt32LSB ) { + stream_.deviceFormat[mode] = RTAUDIO_SINT32; + if ( channelInfo.type == ASIOSTInt32MSB ) stream_.doByteSwap[mode] = true; + } + else if ( channelInfo.type == ASIOSTFloat32MSB || channelInfo.type == ASIOSTFloat32LSB ) { + stream_.deviceFormat[mode] = RTAUDIO_FLOAT32; + if ( channelInfo.type == ASIOSTFloat32MSB ) stream_.doByteSwap[mode] = true; + } + else if ( channelInfo.type == ASIOSTFloat64MSB || channelInfo.type == ASIOSTFloat64LSB ) { + stream_.deviceFormat[mode] = RTAUDIO_FLOAT64; + if ( channelInfo.type == ASIOSTFloat64MSB ) stream_.doByteSwap[mode] = true; + } + else if ( channelInfo.type == ASIOSTInt24MSB || channelInfo.type == ASIOSTInt24LSB ) { + stream_.deviceFormat[mode] = RTAUDIO_SINT24; + if ( channelInfo.type == ASIOSTInt24MSB ) stream_.doByteSwap[mode] = true; + } + + if ( stream_.deviceFormat[mode] == 0 ) { + errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") data format not supported by RtAudio."; + errorText_ = errorStream_.str(); + goto error; + } + + // Set the buffer size. For a duplex stream, this will end up + // setting the buffer size based on the input constraints, which + // should be ok. + long minSize, maxSize, preferSize, granularity; + result = ASIOGetBufferSize( &minSize, &maxSize, &preferSize, &granularity ); + if ( result != ASE_OK ) { + errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error (" << getAsioErrorString( result ) << ") getting buffer size."; + errorText_ = errorStream_.str(); + goto error; + } + + if ( isDuplexInput ) { + // When this is the duplex input (output was opened before), then we have to use the same + // buffersize as the output, because it might use the preferred buffer size, which most + // likely wasn't passed as input to this. The buffer sizes have to be identically anyway, + // So instead of throwing an error, make them equal. The caller uses the reference + // to the "bufferSize" param as usual to set up processing buffers. + + *bufferSize = stream_.bufferSize; + + } else { + if ( *bufferSize == 0 ) *bufferSize = preferSize; + else if ( *bufferSize < (unsigned int) minSize ) *bufferSize = (unsigned int) minSize; + else if ( *bufferSize > (unsigned int) maxSize ) *bufferSize = (unsigned int) maxSize; + else if ( granularity == -1 ) { + // Make sure bufferSize is a power of two. + int log2_of_min_size = 0; + int log2_of_max_size = 0; + + for ( unsigned int i = 0; i < sizeof(long) * 8; i++ ) { + if ( minSize & ((long)1 << i) ) log2_of_min_size = i; + if ( maxSize & ((long)1 << i) ) log2_of_max_size = i; + } + + long min_delta = std::abs( (long)*bufferSize - ((long)1 << log2_of_min_size) ); + int min_delta_num = log2_of_min_size; + + for (int i = log2_of_min_size + 1; i <= log2_of_max_size; i++) { + long current_delta = std::abs( (long)*bufferSize - ((long)1 << i) ); + if (current_delta < min_delta) { + min_delta = current_delta; + min_delta_num = i; + } + } + + *bufferSize = ( (unsigned int)1 << min_delta_num ); + if ( *bufferSize < (unsigned int) minSize ) *bufferSize = (unsigned int) minSize; + else if ( *bufferSize > (unsigned int) maxSize ) *bufferSize = (unsigned int) maxSize; + } + else if ( granularity != 0 ) { + // Set to an even multiple of granularity, rounding up. + *bufferSize = (*bufferSize + granularity-1) / granularity * granularity; + } + } + + /* + // we don't use it anymore, see above! + // Just left it here for the case... + if ( isDuplexInput && stream_.bufferSize != *bufferSize ) { + errorText_ = "RtApiAsio::probeDeviceOpen: input/output buffersize discrepancy!"; + goto error; + } + */ + + stream_.bufferSize = *bufferSize; + stream_.nBuffers = 2; + + if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) stream_.userInterleaved = false; + else stream_.userInterleaved = true; + + // ASIO always uses non-interleaved buffers. + stream_.deviceInterleaved[mode] = false; + + // Allocate, if necessary, our AsioHandle structure for the stream. + if ( handle == 0 ) { + try { + handle = new AsioHandle; + } + catch ( std::bad_alloc& ) { + errorText_ = "RtApiAsio::probeDeviceOpen: error allocating AsioHandle memory."; + goto error; + } + handle->bufferInfos = 0; + + // Create a manual-reset event. + handle->condition = CreateEvent( NULL, // no security + TRUE, // manual-reset + FALSE, // non-signaled initially + NULL ); // unnamed + stream_.apiHandle = (void *) handle; + } + + // Create the ASIO internal buffers. Since RtAudio sets up input + // and output separately, we'll have to dispose of previously + // created output buffers for a duplex stream. + if ( mode == INPUT && stream_.mode == OUTPUT ) { + ASIODisposeBuffers(); + if ( handle->bufferInfos ) free( handle->bufferInfos ); + } + + // Allocate, initialize, and save the bufferInfos in our stream callbackInfo structure. + unsigned int i; + nChannels = stream_.nDeviceChannels[0] + stream_.nDeviceChannels[1]; + handle->bufferInfos = (ASIOBufferInfo *) malloc( nChannels * sizeof(ASIOBufferInfo) ); + if ( handle->bufferInfos == NULL ) { + errorStream_ << "RtApiAsio::probeDeviceOpen: error allocating bufferInfo memory for driver (" << driverName << ")."; + errorText_ = errorStream_.str(); + goto error; + } + + ASIOBufferInfo *infos; + infos = handle->bufferInfos; + for ( i=0; i<stream_.nDeviceChannels[0]; i++, infos++ ) { + infos->isInput = ASIOFalse; + infos->channelNum = i + stream_.channelOffset[0]; + infos->buffers[0] = infos->buffers[1] = 0; + } + for ( i=0; i<stream_.nDeviceChannels[1]; i++, infos++ ) { + infos->isInput = ASIOTrue; + infos->channelNum = i + stream_.channelOffset[1]; + infos->buffers[0] = infos->buffers[1] = 0; + } + + // prepare for callbacks + stream_.sampleRate = sampleRate; + stream_.device[mode] = device; + stream_.mode = isDuplexInput ? DUPLEX : mode; + + // store this class instance before registering callbacks, that are going to use it + asioCallbackInfo = &stream_.callbackInfo; + stream_.callbackInfo.object = (void *) this; + + // Set up the ASIO callback structure and create the ASIO data buffers. + asioCallbacks.bufferSwitch = &bufferSwitch; + asioCallbacks.sampleRateDidChange = &sampleRateChanged; + asioCallbacks.asioMessage = &asioMessages; + asioCallbacks.bufferSwitchTimeInfo = NULL; + result = ASIOCreateBuffers( handle->bufferInfos, nChannels, stream_.bufferSize, &asioCallbacks ); + if ( result != ASE_OK ) { + // Standard method failed. This can happen with strict/misbehaving drivers that return valid buffer size ranges + // but only accept the preferred buffer size as parameter for ASIOCreateBuffers (e.g. Creative's ASIO driver). + // In that case, let's be naïve and try that instead. + *bufferSize = preferSize; + stream_.bufferSize = *bufferSize; + result = ASIOCreateBuffers( handle->bufferInfos, nChannels, stream_.bufferSize, &asioCallbacks ); + } + + if ( result != ASE_OK ) { + errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error (" << getAsioErrorString( result ) << ") creating buffers."; + errorText_ = errorStream_.str(); + goto error; + } + buffersAllocated = true; + stream_.state = STREAM_STOPPED; + + // Set flags for buffer conversion. + stream_.doConvertBuffer[mode] = false; + if ( stream_.userFormat != stream_.deviceFormat[mode] ) + stream_.doConvertBuffer[mode] = true; + if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] && + stream_.nUserChannels[mode] > 1 ) + stream_.doConvertBuffer[mode] = true; + + // Allocate necessary internal buffers + unsigned long bufferBytes; + bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat ); + stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 ); + if ( stream_.userBuffer[mode] == NULL ) { + errorText_ = "RtApiAsio::probeDeviceOpen: error allocating user buffer memory."; + goto error; + } + + if ( stream_.doConvertBuffer[mode] ) { + + bool makeBuffer = true; + bufferBytes = stream_.nDeviceChannels[mode] * formatBytes( stream_.deviceFormat[mode] ); + if ( isDuplexInput && stream_.deviceBuffer ) { + unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] ); + if ( bufferBytes <= bytesOut ) makeBuffer = false; + } + + if ( makeBuffer ) { + bufferBytes *= *bufferSize; + if ( stream_.deviceBuffer ) free( stream_.deviceBuffer ); + stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 ); + if ( stream_.deviceBuffer == NULL ) { + errorText_ = "RtApiAsio::probeDeviceOpen: error allocating device buffer memory."; + goto error; + } + } + } + + // Determine device latencies + long inputLatency, outputLatency; + result = ASIOGetLatencies( &inputLatency, &outputLatency ); + if ( result != ASE_OK ) { + errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error (" << getAsioErrorString( result ) << ") getting latency."; + errorText_ = errorStream_.str(); + error( RtAudioError::WARNING); // warn but don't fail + } + else { + stream_.latency[0] = outputLatency; + stream_.latency[1] = inputLatency; + } + + // Setup the buffer conversion information structure. We don't use + // buffers to do channel offsets, so we override that parameter + // here. + if ( stream_.doConvertBuffer[mode] ) setConvertInfo( mode, 0 ); + + return SUCCESS; + + error: + if ( !isDuplexInput ) { + // the cleanup for error in the duplex input, is done by RtApi::openStream + // So we clean up for single channel only + + if ( buffersAllocated ) + ASIODisposeBuffers(); + + drivers.removeCurrentDriver(); + + if ( handle ) { + CloseHandle( handle->condition ); + if ( handle->bufferInfos ) + free( handle->bufferInfos ); + + delete handle; + stream_.apiHandle = 0; + } + + + if ( stream_.userBuffer[mode] ) { + free( stream_.userBuffer[mode] ); + stream_.userBuffer[mode] = 0; + } + + if ( stream_.deviceBuffer ) { + free( stream_.deviceBuffer ); + stream_.deviceBuffer = 0; + } + } + + return FAILURE; +}//////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////// + +void RtApiAsio :: closeStream() +{ + if ( stream_.state == STREAM_CLOSED ) { + errorText_ = "RtApiAsio::closeStream(): no open stream to close!"; + error( RtAudioError::WARNING ); + return; + } + + if ( stream_.state == STREAM_RUNNING ) { + stream_.state = STREAM_STOPPED; + ASIOStop(); + } + ASIODisposeBuffers(); + drivers.removeCurrentDriver(); + + AsioHandle *handle = (AsioHandle *) stream_.apiHandle; + if ( handle ) { + CloseHandle( handle->condition ); + if ( handle->bufferInfos ) + free( handle->bufferInfos ); + delete handle; + stream_.apiHandle = 0; + } + + for ( int i=0; i<2; i++ ) { + if ( stream_.userBuffer[i] ) { + free( stream_.userBuffer[i] ); + stream_.userBuffer[i] = 0; + } + } + + if ( stream_.deviceBuffer ) { + free( stream_.deviceBuffer ); + stream_.deviceBuffer = 0; + } + + stream_.mode = UNINITIALIZED; + stream_.state = STREAM_CLOSED; +} + +bool stopThreadCalled = false; + +void RtApiAsio :: startStream() +{ + verifyStream(); + if ( stream_.state == STREAM_RUNNING ) { + errorText_ = "RtApiAsio::startStream(): the stream is already running!"; + error( RtAudioError::WARNING ); + return; + } + + #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: + stopThreadCalled = false; + + if ( result == ASE_OK ) return; + error( RtAudioError::SYSTEM_ERROR ); +} + +void RtApiAsio :: stopStream() +{ + verifyStream(); + if ( stream_.state == STREAM_STOPPED ) { + errorText_ = "RtApiAsio::stopStream(): the stream is already stopped!"; + error( RtAudioError::WARNING ); + return; + } + + AsioHandle *handle = (AsioHandle *) stream_.apiHandle; + if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { + if ( handle->drainCounter == 0 ) { + handle->drainCounter = 2; + WaitForSingleObject( handle->condition, INFINITE ); // block until signaled + } + } + + stream_.state = STREAM_STOPPED; + + ASIOError result = ASIOStop(); + if ( result != ASE_OK ) { + errorStream_ << "RtApiAsio::stopStream: error (" << getAsioErrorString( result ) << ") stopping device."; + errorText_ = errorStream_.str(); + } + + if ( result == ASE_OK ) return; + error( RtAudioError::SYSTEM_ERROR ); +} + +void RtApiAsio :: abortStream() +{ + verifyStream(); + if ( stream_.state == STREAM_STOPPED ) { + errorText_ = "RtApiAsio::abortStream(): the stream is already stopped!"; + error( RtAudioError::WARNING ); + return; + } + + // The following lines were commented-out because some behavior was + // noted where the device buffers need to be zeroed to avoid + // continuing sound, even when the device buffers are completely + // disposed. So now, calling abort is the same as calling stop. + // AsioHandle *handle = (AsioHandle *) stream_.apiHandle; + // handle->drainCounter = 2; + stopStream(); +} + +// This function will be called by a spawned thread when the user +// callback function signals that the stream should be stopped or +// aborted. It is necessary to handle it this way because the +// callbackEvent() function must return before the ASIOStop() +// function will return. +static unsigned __stdcall asioStopStream( void *ptr ) +{ + CallbackInfo *info = (CallbackInfo *) ptr; + RtApiAsio *object = (RtApiAsio *) info->object; + + object->stopStream(); + _endthreadex( 0 ); + return 0; +} + +bool RtApiAsio :: callbackEvent( long bufferIndex ) +{ + if ( stream_.state == STREAM_STOPPED || stream_.state == STREAM_STOPPING ) return SUCCESS; + if ( stream_.state == STREAM_CLOSED ) { + errorText_ = "RtApiAsio::callbackEvent(): the stream is closed ... this shouldn't happen!"; + error( RtAudioError::WARNING ); + return FAILURE; + } + + CallbackInfo *info = (CallbackInfo *) &stream_.callbackInfo; + AsioHandle *handle = (AsioHandle *) stream_.apiHandle; + + // Check if we were draining the stream and signal if finished. + if ( handle->drainCounter > 3 ) { + + stream_.state = STREAM_STOPPING; + if ( handle->internalDrain == false ) + SetEvent( handle->condition ); + else { // spawn a thread to stop the stream + unsigned threadId; + stream_.callbackInfo.thread = _beginthreadex( NULL, 0, &asioStopStream, + &stream_.callbackInfo, 0, &threadId ); + } + return SUCCESS; + } + + // Invoke user callback to get fresh output data UNLESS we are + // draining stream. + if ( handle->drainCounter == 0 ) { + RtAudioCallback callback = (RtAudioCallback) info->callback; + double streamTime = getStreamTime(); + RtAudioStreamStatus status = 0; + if ( stream_.mode != INPUT && asioXRun == true ) { + status |= RTAUDIO_OUTPUT_UNDERFLOW; + asioXRun = false; + } + if ( stream_.mode != OUTPUT && asioXRun == true ) { + status |= RTAUDIO_INPUT_OVERFLOW; + asioXRun = false; + } + int cbReturnValue = callback( stream_.userBuffer[0], stream_.userBuffer[1], + stream_.bufferSize, streamTime, status, info->userData ); + if ( cbReturnValue == 2 ) { + stream_.state = STREAM_STOPPING; + handle->drainCounter = 2; + unsigned threadId; + stream_.callbackInfo.thread = _beginthreadex( NULL, 0, &asioStopStream, + &stream_.callbackInfo, 0, &threadId ); + return SUCCESS; + } + else if ( cbReturnValue == 1 ) { + handle->drainCounter = 1; + handle->internalDrain = true; + } + } + + unsigned int nChannels, bufferBytes, i, j; + nChannels = stream_.nDeviceChannels[0] + stream_.nDeviceChannels[1]; + if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { + + bufferBytes = stream_.bufferSize * formatBytes( stream_.deviceFormat[0] ); + + if ( handle->drainCounter > 1 ) { // write zeros to the output stream + + for ( i=0, j=0; i<nChannels; i++ ) { + if ( handle->bufferInfos[i].isInput != ASIOTrue ) + memset( handle->bufferInfos[i].buffers[bufferIndex], 0, bufferBytes ); + } + + } + else if ( stream_.doConvertBuffer[0] ) { + + convertBuffer( stream_.deviceBuffer, stream_.userBuffer[0], stream_.convertInfo[0] ); + if ( stream_.doByteSwap[0] ) + byteSwapBuffer( stream_.deviceBuffer, + stream_.bufferSize * stream_.nDeviceChannels[0], + stream_.deviceFormat[0] ); + + for ( i=0, j=0; i<nChannels; i++ ) { + if ( handle->bufferInfos[i].isInput != ASIOTrue ) + memcpy( handle->bufferInfos[i].buffers[bufferIndex], + &stream_.deviceBuffer[j++*bufferBytes], bufferBytes ); + } + + } + else { + + if ( stream_.doByteSwap[0] ) + byteSwapBuffer( stream_.userBuffer[0], + stream_.bufferSize * stream_.nUserChannels[0], + stream_.userFormat ); + + for ( i=0, j=0; i<nChannels; i++ ) { + if ( handle->bufferInfos[i].isInput != ASIOTrue ) + memcpy( handle->bufferInfos[i].buffers[bufferIndex], + &stream_.userBuffer[0][bufferBytes*j++], bufferBytes ); + } + + } + } + + // Don't bother draining input + if ( handle->drainCounter ) { + handle->drainCounter++; + goto unlock; + } + + if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) { + + bufferBytes = stream_.bufferSize * formatBytes(stream_.deviceFormat[1]); + + if (stream_.doConvertBuffer[1]) { + + // Always interleave ASIO input data. + for ( i=0, j=0; i<nChannels; i++ ) { + if ( handle->bufferInfos[i].isInput == ASIOTrue ) + memcpy( &stream_.deviceBuffer[j++*bufferBytes], + handle->bufferInfos[i].buffers[bufferIndex], + bufferBytes ); + } + + if ( stream_.doByteSwap[1] ) + byteSwapBuffer( stream_.deviceBuffer, + stream_.bufferSize * stream_.nDeviceChannels[1], + stream_.deviceFormat[1] ); + convertBuffer( stream_.userBuffer[1], stream_.deviceBuffer, stream_.convertInfo[1] ); + + } + else { + for ( i=0, j=0; i<nChannels; i++ ) { + if ( handle->bufferInfos[i].isInput == ASIOTrue ) { + memcpy( &stream_.userBuffer[1][bufferBytes*j++], + handle->bufferInfos[i].buffers[bufferIndex], + bufferBytes ); + } + } + + if ( stream_.doByteSwap[1] ) + byteSwapBuffer( stream_.userBuffer[1], + stream_.bufferSize * stream_.nUserChannels[1], + stream_.userFormat ); + } + } + + unlock: + // The following call was suggested by Malte Clasen. While the API + // documentation indicates it should not be required, some device + // drivers apparently do not function correctly without it. + ASIOOutputReady(); + + RtApi::tickStreamTime(); + return SUCCESS; +} + +static void sampleRateChanged( ASIOSampleRate sRate ) +{ + // The ASIO documentation says that this usually only happens during + // external sync. Audio processing is not stopped by the driver, + // actual sample rate might not have even changed, maybe only the + // sample rate status of an AES/EBU or S/PDIF digital input at the + // audio device. + + RtApi *object = (RtApi *) asioCallbackInfo->object; + try { + object->stopStream(); + } + catch ( RtAudioError &exception ) { + std::cerr << "\nRtApiAsio: sampleRateChanged() error (" << exception.getMessage() << ")!\n" << std::endl; + return; + } + + std::cerr << "\nRtApiAsio: driver reports sample rate changed to " << sRate << " ... stream stopped!!!\n" << std::endl; +} + +static long asioMessages( long selector, long value, void* /*message*/, double* /*opt*/ ) +{ + long ret = 0; + + switch( selector ) { + case kAsioSelectorSupported: + if ( value == kAsioResetRequest + || value == kAsioEngineVersion + || value == kAsioResyncRequest + || value == kAsioLatenciesChanged + // The following three were added for ASIO 2.0, you don't + // necessarily have to support them. + || value == kAsioSupportsTimeInfo + || value == kAsioSupportsTimeCode + || value == kAsioSupportsInputMonitor) + ret = 1L; + break; + case kAsioResetRequest: + // Defer the task and perform the reset of the driver during the + // next "safe" situation. You cannot reset the driver right now, + // as this code is called from the driver. Reset the driver is + // done by completely destruct is. I.e. ASIOStop(), + // ASIODisposeBuffers(), Destruction Afterwards you initialize the + // driver again. + std::cerr << "\nRtApiAsio: driver reset requested!!!" << std::endl; + ret = 1L; + break; + case kAsioResyncRequest: + // This informs the application that the driver encountered some + // non-fatal data loss. It is used for synchronization purposes + // of different media. Added mainly to work around the Win16Mutex + // problems in Windows 95/98 with the Windows Multimedia system, + // which could lose data because the Mutex was held too long by + // another thread. However a driver can issue it in other + // situations, too. + // std::cerr << "\nRtApiAsio: driver resync requested!!!" << std::endl; + asioXRun = true; + ret = 1L; + break; + case kAsioLatenciesChanged: + // This will inform the host application that the drivers were + // latencies changed. Beware, it this does not mean that the + // buffer sizes have changed! You might need to update internal + // delay data. + std::cerr << "\nRtApiAsio: driver latency may have changed!!!" << std::endl; + ret = 1L; + break; + case kAsioEngineVersion: + // Return the supported ASIO version of the host application. If + // a host application does not implement this selector, ASIO 1.0 + // is assumed by the driver. + ret = 2L; + break; + case kAsioSupportsTimeInfo: + // Informs the driver whether the + // asioCallbacks.bufferSwitchTimeInfo() callback is supported. + // For compatibility with ASIO 1.0 drivers the host application + // should always support the "old" bufferSwitch method, too. + ret = 0; + break; + case kAsioSupportsTimeCode: + // Informs the driver whether application is interested in time + // code info. If an application does not need to know about time + // code, the driver has less work to do. + ret = 0; + break; + } + return ret; +} + +static const char* getAsioErrorString( ASIOError result ) +{ + struct Messages + { + ASIOError value; + const char*message; + }; + + static const Messages m[] = + { + { ASE_NotPresent, "Hardware input or output is not present or available." }, + { ASE_HWMalfunction, "Hardware is malfunctioning." }, + { ASE_InvalidParameter, "Invalid input parameter." }, + { ASE_InvalidMode, "Invalid mode." }, + { ASE_SPNotAdvancing, "Sample position not advancing." }, + { ASE_NoClock, "Sample clock or rate cannot be determined or is not present." }, + { ASE_NoMemory, "Not enough memory to complete the request." } + }; + + for ( unsigned int i = 0; i < sizeof(m)/sizeof(m[0]); ++i ) + if ( m[i].value == result ) return m[i].message; + + return "Unknown error."; +} + +//******************** End of __WINDOWS_ASIO__ *********************// +#endif + + +#if defined(__WINDOWS_WASAPI__) // Windows WASAPI API + +// Authored by Marcus Tomlinson <themarcustomlinson@gmail.com>, April 2014 +// - Introduces support for the Windows WASAPI API +// - Aims to deliver bit streams to and from hardware at the lowest possible latency, via the absolute minimum buffer sizes required +// - Provides flexible stream configuration to an otherwise strict and inflexible WASAPI interface +// - Includes automatic internal conversion of sample rate and buffer size between hardware and the user + +#ifndef INITGUID + #define INITGUID +#endif + +#include <mfapi.h> +#include <mferror.h> +#include <mfplay.h> +#include <mftransform.h> +#include <wmcodecdsp.h> + +#include <audioclient.h> +#include <avrt.h> +#include <mmdeviceapi.h> +#include <functiondiscoverykeys_devpkey.h> + +#ifndef MF_E_TRANSFORM_NEED_MORE_INPUT + #define MF_E_TRANSFORM_NEED_MORE_INPUT _HRESULT_TYPEDEF_(0xc00d6d72) +#endif + +#ifndef MFSTARTUP_NOSOCKET + #define MFSTARTUP_NOSOCKET 0x1 +#endif + +#ifdef _MSC_VER + #pragma comment( lib, "ksuser" ) + #pragma comment( lib, "mfplat.lib" ) + #pragma comment( lib, "mfuuid.lib" ) + #pragma comment( lib, "wmcodecdspuuid" ) +#endif + +//============================================================================= + +#define SAFE_RELEASE( objectPtr )\ +if ( objectPtr )\ +{\ + objectPtr->Release();\ + objectPtr = NULL;\ +} + +typedef HANDLE ( __stdcall *TAvSetMmThreadCharacteristicsPtr )( LPCWSTR TaskName, LPDWORD TaskIndex ); + +//----------------------------------------------------------------------------- + +// WASAPI dictates stream sample rate, format, channel count, and in some cases, buffer size. +// Therefore we must perform all necessary conversions to user buffers in order to satisfy these +// requirements. WasapiBuffer ring buffers are used between HwIn->UserIn and UserOut->HwOut to +// provide intermediate storage for read / write synchronization. +class WasapiBuffer +{ +public: + WasapiBuffer() + : buffer_( NULL ), + bufferSize_( 0 ), + inIndex_( 0 ), + outIndex_( 0 ) {} + + ~WasapiBuffer() { + free( buffer_ ); + } + + // sets the length of the internal ring buffer + void setBufferSize( unsigned int bufferSize, unsigned int formatBytes ) { + free( buffer_ ); + + buffer_ = ( char* ) calloc( bufferSize, formatBytes ); + + bufferSize_ = bufferSize; + inIndex_ = 0; + outIndex_ = 0; + } + + // attempt to push a buffer into the ring buffer at the current "in" index + bool pushBuffer( char* buffer, unsigned int bufferSize, RtAudioFormat format ) + { + if ( !buffer || // incoming buffer is NULL + bufferSize == 0 || // incoming buffer has no data + bufferSize > bufferSize_ ) // incoming buffer too large + { + return false; + } + + unsigned int relOutIndex = outIndex_; + unsigned int inIndexEnd = inIndex_ + bufferSize; + if ( relOutIndex < inIndex_ && inIndexEnd >= bufferSize_ ) { + relOutIndex += bufferSize_; + } + + // 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() +{ + if ( stream_.state != STREAM_CLOSED ) + closeStream(); + + SAFE_RELEASE( deviceEnumerator_ ); + + // If this object previously called CoInitialize() + if ( coInitialized_ ) + CoUninitialize(); +} + +//============================================================================= + +unsigned int RtApiWasapi::getDeviceCount( void ) +{ + unsigned int captureDeviceCount = 0; + unsigned int renderDeviceCount = 0; + + IMMDeviceCollection* captureDevices = NULL; + IMMDeviceCollection* renderDevices = NULL; + + if ( !deviceEnumerator_ ) + return 0; + + // Count capture devices + errorText_.clear(); + HRESULT hr = deviceEnumerator_->EnumAudioEndpoints( eCapture, DEVICE_STATE_ACTIVE, &captureDevices ); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::getDeviceCount: Unable to retrieve capture device collection."; + goto Exit; + } + + hr = captureDevices->GetCount( &captureDeviceCount ); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::getDeviceCount: Unable to retrieve capture device count."; + goto Exit; + } + + // Count render devices + hr = deviceEnumerator_->EnumAudioEndpoints( eRender, DEVICE_STATE_ACTIVE, &renderDevices ); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::getDeviceCount: Unable to retrieve render device collection."; + goto Exit; + } + + hr = renderDevices->GetCount( &renderDeviceCount ); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::getDeviceCount: Unable to retrieve render device count."; + goto Exit; + } + +Exit: + // release all references + SAFE_RELEASE( captureDevices ); + SAFE_RELEASE( renderDevices ); + + if ( errorText_.empty() ) + return captureDeviceCount + renderDeviceCount; + + error( RtAudioError::DRIVER_ERROR ); + return 0; +} + +//----------------------------------------------------------------------------- + +RtAudio::DeviceInfo RtApiWasapi::getDeviceInfo( unsigned int device ) +{ + RtAudio::DeviceInfo info; + unsigned int captureDeviceCount = 0; + unsigned int renderDeviceCount = 0; + std::string defaultDeviceName; + bool isCaptureDevice = false; + + PROPVARIANT deviceNameProp; + PROPVARIANT defaultDeviceNameProp; + + IMMDeviceCollection* captureDevices = NULL; + IMMDeviceCollection* renderDevices = NULL; + IMMDevice* devicePtr = NULL; + IMMDevice* defaultDevicePtr = NULL; + IAudioClient* audioClient = NULL; + IPropertyStore* devicePropStore = NULL; + IPropertyStore* defaultDevicePropStore = NULL; + + WAVEFORMATEX* deviceFormat = NULL; + WAVEFORMATEX* closestMatchFormat = NULL; + + // probed + info.probed = false; + + // Count capture devices + errorText_.clear(); + RtAudioError::Type errorType = RtAudioError::DRIVER_ERROR; + HRESULT hr = deviceEnumerator_->EnumAudioEndpoints( eCapture, DEVICE_STATE_ACTIVE, &captureDevices ); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve capture device collection."; + goto Exit; + } + + hr = captureDevices->GetCount( &captureDeviceCount ); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve capture device count."; + goto Exit; + } + + // Count render devices + hr = deviceEnumerator_->EnumAudioEndpoints( eRender, DEVICE_STATE_ACTIVE, &renderDevices ); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve render device collection."; + goto Exit; + } + + hr = renderDevices->GetCount( &renderDeviceCount ); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve render device count."; + goto Exit; + } + + // validate device index + if ( device >= captureDeviceCount + renderDeviceCount ) { + errorText_ = "RtApiWasapi::getDeviceInfo: Invalid device index."; + errorType = RtAudioError::INVALID_USE; + goto Exit; + } + + // determine whether index falls within capture or render devices + if ( device >= renderDeviceCount ) { + hr = captureDevices->Item( device - renderDeviceCount, &devicePtr ); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve capture device handle."; + goto Exit; + } + isCaptureDevice = true; + } + else { + hr = renderDevices->Item( device, &devicePtr ); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve render device handle."; + goto Exit; + } + isCaptureDevice = false; + } + + // get default device name + if ( isCaptureDevice ) { + hr = deviceEnumerator_->GetDefaultAudioEndpoint( eCapture, eConsole, &defaultDevicePtr ); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve default capture device handle."; + goto Exit; + } + } + else { + hr = deviceEnumerator_->GetDefaultAudioEndpoint( eRender, eConsole, &defaultDevicePtr ); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve default render device handle."; + goto Exit; + } + } + + hr = defaultDevicePtr->OpenPropertyStore( STGM_READ, &defaultDevicePropStore ); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::getDeviceInfo: Unable to open default device property store."; + goto Exit; + } + PropVariantInit( &defaultDeviceNameProp ); + + hr = defaultDevicePropStore->GetValue( PKEY_Device_FriendlyName, &defaultDeviceNameProp ); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve default device property: PKEY_Device_FriendlyName."; + goto Exit; + } + + defaultDeviceName = convertCharPointerToStdString(defaultDeviceNameProp.pwszVal); + + // name + hr = devicePtr->OpenPropertyStore( STGM_READ, &devicePropStore ); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::getDeviceInfo: Unable to open device property store."; + goto Exit; + } + + PropVariantInit( &deviceNameProp ); + + hr = devicePropStore->GetValue( PKEY_Device_FriendlyName, &deviceNameProp ); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve device property: PKEY_Device_FriendlyName."; + goto Exit; + } + + info.name =convertCharPointerToStdString(deviceNameProp.pwszVal); + + // is default + if ( isCaptureDevice ) { + info.isDefaultInput = info.name == defaultDeviceName; + info.isDefaultOutput = false; + } + else { + info.isDefaultInput = false; + info.isDefaultOutput = info.name == defaultDeviceName; + } + + // channel count + hr = devicePtr->Activate( __uuidof( IAudioClient ), CLSCTX_ALL, NULL, ( void** ) &audioClient ); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve device audio client."; + goto Exit; + } + + hr = audioClient->GetMixFormat( &deviceFormat ); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve device mix format."; + goto Exit; + } + + if ( isCaptureDevice ) { + info.inputChannels = deviceFormat->nChannels; + info.outputChannels = 0; + info.duplexChannels = 0; + } + else { + info.inputChannels = 0; + info.outputChannels = deviceFormat->nChannels; + info.duplexChannels = 0; + } + + // sample rates + info.sampleRates.clear(); + + // allow support for all sample rates as we have a built-in sample rate converter + for ( unsigned int i = 0; i < MAX_SAMPLE_RATES; i++ ) { + info.sampleRates.push_back( SAMPLE_RATES[i] ); + } + info.preferredSampleRate = deviceFormat->nSamplesPerSec; + + // native format + info.nativeFormats = 0; + + if ( deviceFormat->wFormatTag == WAVE_FORMAT_IEEE_FLOAT || + ( deviceFormat->wFormatTag == WAVE_FORMAT_EXTENSIBLE && + ( ( WAVEFORMATEXTENSIBLE* ) deviceFormat )->SubFormat == KSDATAFORMAT_SUBTYPE_IEEE_FLOAT ) ) + { + if ( deviceFormat->wBitsPerSample == 32 ) { + info.nativeFormats |= RTAUDIO_FLOAT32; + } + else if ( deviceFormat->wBitsPerSample == 64 ) { + info.nativeFormats |= RTAUDIO_FLOAT64; + } + } + else if ( deviceFormat->wFormatTag == WAVE_FORMAT_PCM || + ( deviceFormat->wFormatTag == WAVE_FORMAT_EXTENSIBLE && + ( ( WAVEFORMATEXTENSIBLE* ) deviceFormat )->SubFormat == KSDATAFORMAT_SUBTYPE_PCM ) ) + { + if ( deviceFormat->wBitsPerSample == 8 ) { + info.nativeFormats |= RTAUDIO_SINT8; + } + else if ( deviceFormat->wBitsPerSample == 16 ) { + info.nativeFormats |= RTAUDIO_SINT16; + } + else if ( deviceFormat->wBitsPerSample == 24 ) { + info.nativeFormats |= RTAUDIO_SINT24; + } + else if ( deviceFormat->wBitsPerSample == 32 ) { + info.nativeFormats |= RTAUDIO_SINT32; + } + } + + // probed + info.probed = true; + +Exit: + // release all references + PropVariantClear( &deviceNameProp ); + PropVariantClear( &defaultDeviceNameProp ); + + SAFE_RELEASE( captureDevices ); + SAFE_RELEASE( renderDevices ); + SAFE_RELEASE( devicePtr ); + SAFE_RELEASE( defaultDevicePtr ); + SAFE_RELEASE( audioClient ); + SAFE_RELEASE( devicePropStore ); + SAFE_RELEASE( defaultDevicePropStore ); + + CoTaskMemFree( deviceFormat ); + CoTaskMemFree( closestMatchFormat ); + + if ( !errorText_.empty() ) + error( errorType ); + return info; +} + +//----------------------------------------------------------------------------- + +unsigned int RtApiWasapi::getDefaultOutputDevice( void ) +{ + for ( unsigned int i = 0; i < getDeviceCount(); i++ ) { + if ( getDeviceInfo( i ).isDefaultOutput ) { + return i; + } + } + + return 0; +} + +//----------------------------------------------------------------------------- + +unsigned int RtApiWasapi::getDefaultInputDevice( void ) +{ + for ( unsigned int i = 0; i < getDeviceCount(); i++ ) { + if ( getDeviceInfo( i ).isDefaultInput ) { + return i; + } + } + + return 0; +} + +//----------------------------------------------------------------------------- + +void RtApiWasapi::closeStream( void ) +{ + if ( stream_.state == STREAM_CLOSED ) { + errorText_ = "RtApiWasapi::closeStream: No open stream to close."; + error( RtAudioError::WARNING ); + return; + } + + if ( stream_.state != STREAM_STOPPED ) + stopStream(); + + // clean up stream memory + SAFE_RELEASE( ( ( WasapiHandle* ) stream_.apiHandle )->captureAudioClient ) + SAFE_RELEASE( ( ( WasapiHandle* ) stream_.apiHandle )->renderAudioClient ) + + SAFE_RELEASE( ( ( WasapiHandle* ) stream_.apiHandle )->captureClient ) + SAFE_RELEASE( ( ( WasapiHandle* ) stream_.apiHandle )->renderClient ) + + if ( ( ( WasapiHandle* ) stream_.apiHandle )->captureEvent ) + CloseHandle( ( ( WasapiHandle* ) stream_.apiHandle )->captureEvent ); + + if ( ( ( WasapiHandle* ) stream_.apiHandle )->renderEvent ) + CloseHandle( ( ( WasapiHandle* ) stream_.apiHandle )->renderEvent ); + + delete ( WasapiHandle* ) stream_.apiHandle; + stream_.apiHandle = NULL; + + for ( int i = 0; i < 2; i++ ) { + if ( stream_.userBuffer[i] ) { + free( stream_.userBuffer[i] ); + stream_.userBuffer[i] = 0; + } + } + + if ( stream_.deviceBuffer ) { + free( stream_.deviceBuffer ); + stream_.deviceBuffer = 0; + } + + // update stream state + stream_.state = STREAM_CLOSED; +} + +//----------------------------------------------------------------------------- + +void RtApiWasapi::startStream( void ) +{ + verifyStream(); + + if ( stream_.state == STREAM_RUNNING ) { + errorText_ = "RtApiWasapi::startStream: The stream is already running."; + error( RtAudioError::WARNING ); + return; + } + + #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."; + 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 ); + + // 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 ); + } + + // close thread handle + if ( stream_.callbackInfo.thread && !CloseHandle( ( void* ) stream_.callbackInfo.thread ) ) { + errorText_ = "RtApiWasapi::abortStream: Unable to close callback thread."; + error( RtAudioError::THREAD_ERROR ); + return; + } + + stream_.callbackInfo.thread = (ThreadHandle) NULL; +} + +//----------------------------------------------------------------------------- + +bool RtApiWasapi::probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels, + unsigned int firstChannel, unsigned int sampleRate, + RtAudioFormat format, unsigned int* bufferSize, + RtAudio::StreamOptions* options ) +{ + bool methodResult = FAILURE; + unsigned int captureDeviceCount = 0; + unsigned int renderDeviceCount = 0; + + IMMDeviceCollection* captureDevices = NULL; + IMMDeviceCollection* renderDevices = NULL; + IMMDevice* devicePtr = NULL; + WAVEFORMATEX* deviceFormat = NULL; + unsigned int bufferBytes; + stream_.state = STREAM_STOPPED; + + // create API Handle if not already created + if ( !stream_.apiHandle ) + stream_.apiHandle = ( void* ) new WasapiHandle(); + + // Count capture devices + errorText_.clear(); + RtAudioError::Type errorType = RtAudioError::DRIVER_ERROR; + HRESULT hr = deviceEnumerator_->EnumAudioEndpoints( eCapture, DEVICE_STATE_ACTIVE, &captureDevices ); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve capture device collection."; + goto Exit; + } + + hr = captureDevices->GetCount( &captureDeviceCount ); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve capture device count."; + goto Exit; + } + + // Count render devices + hr = deviceEnumerator_->EnumAudioEndpoints( eRender, DEVICE_STATE_ACTIVE, &renderDevices ); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve render device collection."; + goto Exit; + } + + hr = renderDevices->GetCount( &renderDeviceCount ); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve render device count."; + goto Exit; + } + + // validate device index + if ( device >= captureDeviceCount + renderDeviceCount ) { + errorType = RtAudioError::INVALID_USE; + errorText_ = "RtApiWasapi::probeDeviceOpen: Invalid device index."; + goto Exit; + } + + // if device index falls within capture devices + if ( device >= renderDeviceCount ) { + if ( mode != INPUT ) { + errorType = RtAudioError::INVALID_USE; + errorText_ = "RtApiWasapi::probeDeviceOpen: Capture device selected as output device."; + goto Exit; + } + + // retrieve captureAudioClient from devicePtr + IAudioClient*& captureAudioClient = ( ( WasapiHandle* ) stream_.apiHandle )->captureAudioClient; + + hr = captureDevices->Item( device - renderDeviceCount, &devicePtr ); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve capture device handle."; + goto Exit; + } + + hr = devicePtr->Activate( __uuidof( IAudioClient ), CLSCTX_ALL, + NULL, ( void** ) &captureAudioClient ); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve capture device audio client."; + goto Exit; + } + + hr = captureAudioClient->GetMixFormat( &deviceFormat ); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve capture device mix format."; + goto Exit; + } + + stream_.nDeviceChannels[mode] = deviceFormat->nChannels; + captureAudioClient->GetStreamLatency( ( long long* ) &stream_.latency[mode] ); + } + + // if device index falls within render devices and is configured for loopback + if ( device < renderDeviceCount && mode == INPUT ) + { + // if renderAudioClient is not initialised, initialise it now + IAudioClient*& renderAudioClient = ( ( WasapiHandle* ) stream_.apiHandle )->renderAudioClient; + if ( !renderAudioClient ) + { + probeDeviceOpen( device, OUTPUT, channels, firstChannel, sampleRate, format, bufferSize, options ); + } + + // retrieve captureAudioClient from devicePtr + IAudioClient*& captureAudioClient = ( ( WasapiHandle* ) stream_.apiHandle )->captureAudioClient; + + hr = renderDevices->Item( device, &devicePtr ); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve render device handle."; + goto Exit; + } + + hr = devicePtr->Activate( __uuidof( IAudioClient ), CLSCTX_ALL, + NULL, ( void** ) &captureAudioClient ); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve render device audio client."; + goto Exit; + } + + hr = captureAudioClient->GetMixFormat( &deviceFormat ); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve render device mix format."; + goto Exit; + } + + stream_.nDeviceChannels[mode] = deviceFormat->nChannels; + captureAudioClient->GetStreamLatency( ( long long* ) &stream_.latency[mode] ); + } + + // if device index falls within render devices and is configured for output + if ( device < renderDeviceCount && mode == OUTPUT ) + { + // if renderAudioClient is already initialised, don't initialise it again + IAudioClient*& renderAudioClient = ( ( WasapiHandle* ) stream_.apiHandle )->renderAudioClient; + if ( renderAudioClient ) + { + methodResult = SUCCESS; + goto Exit; + } + + hr = renderDevices->Item( device, &devicePtr ); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve render device handle."; + goto Exit; + } + + hr = devicePtr->Activate( __uuidof( IAudioClient ), CLSCTX_ALL, + NULL, ( void** ) &renderAudioClient ); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve render device audio client."; + goto Exit; + } + + hr = renderAudioClient->GetMixFormat( &deviceFormat ); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve render device mix format."; + goto Exit; + } + + stream_.nDeviceChannels[mode] = deviceFormat->nChannels; + renderAudioClient->GetStreamLatency( ( long long* ) &stream_.latency[mode] ); + } + + // fill stream data + if ( ( stream_.mode == OUTPUT && mode == INPUT ) || + ( stream_.mode == INPUT && mode == OUTPUT ) ) { + stream_.mode = DUPLEX; + } + else { + stream_.mode = mode; + } + + stream_.device[mode] = device; + stream_.doByteSwap[mode] = false; + stream_.sampleRate = sampleRate; + stream_.bufferSize = *bufferSize; + stream_.nBuffers = 1; + stream_.nUserChannels[mode] = channels; + stream_.channelOffset[mode] = firstChannel; + stream_.userFormat = format; + stream_.deviceFormat[mode] = getDeviceInfo( device ).nativeFormats; + + if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) + stream_.userInterleaved = false; + else + stream_.userInterleaved = true; + stream_.deviceInterleaved[mode] = true; + + // Set flags for buffer conversion. + stream_.doConvertBuffer[mode] = false; + if ( stream_.userFormat != stream_.deviceFormat[mode] || + stream_.nUserChannels[0] != stream_.nDeviceChannels[0] || + stream_.nUserChannels[1] != stream_.nDeviceChannels[1] ) + stream_.doConvertBuffer[mode] = true; + else if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] && + stream_.nUserChannels[mode] > 1 ) + stream_.doConvertBuffer[mode] = true; + + if ( stream_.doConvertBuffer[mode] ) + setConvertInfo( mode, 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 = RtAudioError::MEMORY_ERROR; + errorText_ = "RtApiWasapi::probeDeviceOpen: Error allocating user buffer memory."; + goto Exit; + } + + if ( options && options->flags & RTAUDIO_SCHEDULE_REALTIME ) + stream_.callbackInfo.priority = 15; + else + stream_.callbackInfo.priority = 0; + + ///! TODO: RTAUDIO_MINIMIZE_LATENCY // Provide stream buffers directly to callback + ///! TODO: RTAUDIO_HOG_DEVICE // Exclusive mode + + methodResult = SUCCESS; + +Exit: + //clean up + SAFE_RELEASE( captureDevices ); + SAFE_RELEASE( renderDevices ); + SAFE_RELEASE( devicePtr ); + CoTaskMemFree( deviceFormat ); + + // if method failed, close the stream + if ( methodResult == FAILURE ) + closeStream(); + + if ( !errorText_.empty() ) + error( errorType ); + return methodResult; +} + +//============================================================================= + +DWORD WINAPI RtApiWasapi::runWasapiThread( void* wasapiPtr ) +{ + if ( wasapiPtr ) + ( ( RtApiWasapi* ) wasapiPtr )->wasapiThread(); + + return 0; +} + +DWORD WINAPI RtApiWasapi::stopWasapiThread( void* wasapiPtr ) +{ + if ( wasapiPtr ) + ( ( RtApiWasapi* ) wasapiPtr )->stopStream(); + + return 0; +} + +DWORD WINAPI RtApiWasapi::abortWasapiThread( void* wasapiPtr ) +{ + if ( wasapiPtr ) + ( ( RtApiWasapi* ) wasapiPtr )->abortStream(); + + return 0; +} + +//----------------------------------------------------------------------------- + +void RtApiWasapi::wasapiThread() +{ + // as this is a new thread, we must CoInitialize it + CoInitialize( NULL ); + + HRESULT hr; + + IAudioClient* captureAudioClient = ( ( WasapiHandle* ) stream_.apiHandle )->captureAudioClient; + IAudioClient* renderAudioClient = ( ( WasapiHandle* ) stream_.apiHandle )->renderAudioClient; + IAudioCaptureClient* captureClient = ( ( WasapiHandle* ) stream_.apiHandle )->captureClient; + IAudioRenderClient* renderClient = ( ( WasapiHandle* ) stream_.apiHandle )->renderClient; + HANDLE captureEvent = ( ( WasapiHandle* ) stream_.apiHandle )->captureEvent; + HANDLE renderEvent = ( ( WasapiHandle* ) stream_.apiHandle )->renderEvent; + + WAVEFORMATEX* captureFormat = NULL; + WAVEFORMATEX* renderFormat = NULL; + float captureSrRatio = 0.0f; + float renderSrRatio = 0.0f; + WasapiBuffer captureBuffer; + WasapiBuffer renderBuffer; + WasapiResampler* captureResampler = NULL; + WasapiResampler* renderResampler = NULL; + + // declare local stream variables + RtAudioCallback callback = ( RtAudioCallback ) stream_.callbackInfo.callback; + BYTE* streamBuffer = NULL; + DWORD captureFlags = 0; + unsigned int bufferFrameCount = 0; + unsigned int numFramesPadding = 0; + unsigned int convBufferSize = 0; + bool loopbackEnabled = stream_.device[INPUT] == stream_.device[OUTPUT]; + bool callbackPushed = true; + bool callbackPulled = false; + bool callbackStopped = false; + int callbackResult = 0; + + // convBuffer is used to store converted buffers between WASAPI and the user + char* convBuffer = NULL; + unsigned int convBuffSize = 0; + unsigned int deviceBuffSize = 0; + + std::string errorText; + 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 ) (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 ) { + hr = captureAudioClient->Initialize( AUDCLNT_SHAREMODE_SHARED, + loopbackEnabled ? AUDCLNT_STREAMFLAGS_LOOPBACK : AUDCLNT_STREAMFLAGS_EVENTCALLBACK, + 0, + 0, + captureFormat, + NULL ); + if ( FAILED( hr ) ) { + errorText = "RtApiWasapi::wasapiThread: Unable to initialize capture audio client."; + goto Exit; + } + + hr = captureAudioClient->GetService( __uuidof( IAudioCaptureClient ), + ( void** ) &captureClient ); + if ( FAILED( hr ) ) { + errorText = "RtApiWasapi::wasapiThread: Unable to retrieve capture client handle."; + goto Exit; + } + + // don't configure captureEvent if in loopback mode + if ( !loopbackEnabled ) + { + // configure captureEvent to trigger on every available capture buffer + captureEvent = CreateEvent( NULL, FALSE, FALSE, NULL ); + if ( !captureEvent ) { + errorType = RtAudioError::SYSTEM_ERROR; + errorText = "RtApiWasapi::wasapiThread: Unable to create capture event."; + goto Exit; + } + + hr = captureAudioClient->SetEventHandle( captureEvent ); + if ( FAILED( hr ) ) { + errorText = "RtApiWasapi::wasapiThread: Unable to set capture event handle."; + goto Exit; + } + + ( ( WasapiHandle* ) stream_.apiHandle )->captureEvent = captureEvent; + } + + ( ( WasapiHandle* ) stream_.apiHandle )->captureClient = captureClient; + + // 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 ) { + hr = renderAudioClient->Initialize( AUDCLNT_SHAREMODE_SHARED, + AUDCLNT_STREAMFLAGS_EVENTCALLBACK, + 0, + 0, + renderFormat, + NULL ); + if ( FAILED( hr ) ) { + errorText = "RtApiWasapi::wasapiThread: Unable to initialize render audio client."; + goto Exit; + } + + hr = renderAudioClient->GetService( __uuidof( IAudioRenderClient ), + ( void** ) &renderClient ); + if ( FAILED( hr ) ) { + errorText = "RtApiWasapi::wasapiThread: Unable to retrieve render client handle."; + goto Exit; + } + + // configure renderEvent to trigger on every available render buffer + renderEvent = CreateEvent( NULL, FALSE, FALSE, NULL ); + if ( !renderEvent ) { + errorType = RtAudioError::SYSTEM_ERROR; + errorText = "RtApiWasapi::wasapiThread: Unable to create render event."; + goto Exit; + } + + hr = renderAudioClient->SetEventHandle( renderEvent ); + if ( FAILED( hr ) ) { + errorText = "RtApiWasapi::wasapiThread: Unable to set render event handle."; + goto Exit; + } + + ( ( WasapiHandle* ) stream_.apiHandle )->renderClient = renderClient; + ( ( WasapiHandle* ) stream_.apiHandle )->renderEvent = renderEvent; + + // 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 = ( size_t ) ( ceilf( stream_.bufferSize * captureSrRatio ) ) * stream_.nDeviceChannels[INPUT] * formatBytes( stream_.deviceFormat[INPUT] ); + deviceBuffSize = stream_.bufferSize * stream_.nDeviceChannels[INPUT] * formatBytes( stream_.deviceFormat[INPUT] ); + } + else if ( stream_.mode == OUTPUT ) + { + convBuffSize = ( size_t ) ( ceilf( stream_.bufferSize * renderSrRatio ) ) * stream_.nDeviceChannels[OUTPUT] * formatBytes( stream_.deviceFormat[OUTPUT] ); + deviceBuffSize = stream_.bufferSize * stream_.nDeviceChannels[OUTPUT] * formatBytes( stream_.deviceFormat[OUTPUT] ); + } + else if ( stream_.mode == DUPLEX ) + { + convBuffSize = std::max( ( size_t ) ( ceilf( stream_.bufferSize * captureSrRatio ) ) * stream_.nDeviceChannels[INPUT] * formatBytes( stream_.deviceFormat[INPUT] ), + ( size_t ) ( ceilf( stream_.bufferSize * renderSrRatio ) ) * stream_.nDeviceChannels[OUTPUT] * formatBytes( stream_.deviceFormat[OUTPUT] ) ); + deviceBuffSize = std::max( stream_.bufferSize * stream_.nDeviceChannels[INPUT] * formatBytes( stream_.deviceFormat[INPUT] ), + stream_.bufferSize * stream_.nDeviceChannels[OUTPUT] * formatBytes( stream_.deviceFormat[OUTPUT] ) ); + } + + convBuffSize *= 2; // allow overflow for *SrRatio remainders + convBuffer = ( char* ) calloc( convBuffSize, 1 ); + stream_.deviceBuffer = ( char* ) calloc( deviceBuffSize, 1 ); + if ( !convBuffer || !stream_.deviceBuffer ) { + errorType = RtAudioError::MEMORY_ERROR; + errorText = "RtApiWasapi::wasapiThread: Error allocating device buffer memory."; + goto Exit; + } + + // stream process loop + while ( stream_.state != STREAM_STOPPING ) { + if ( !callbackPulled ) { + // Callback Input + // ============== + // 1. Pull callback buffer from inputBuffer + // 2. If 1. was successful: Convert callback buffer to user sample rate and channel count + // Convert callback buffer to user format + + if ( captureAudioClient ) + { + int samplesToPull = ( unsigned int ) floorf( stream_.bufferSize * captureSrRatio ); + + 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 = RtAudioError::THREAD_ERROR; + errorText = "RtApiWasapi::wasapiThread: Unable to instantiate stream stop thread."; + goto Exit; + } + else if ( !CloseHandle( threadHandle ) ) { + errorType = RtAudioError::THREAD_ERROR; + errorText = "RtApiWasapi::wasapiThread: Unable to close stream stop thread handle."; + goto Exit; + } + + callbackStopped = true; + } + else if ( callbackResult == 2 ) { + // instantiate a thread to stop this thread + HANDLE threadHandle = CreateThread( NULL, 0, abortWasapiThread, this, 0, NULL ); + if ( !threadHandle ) { + errorType = RtAudioError::THREAD_ERROR; + errorText = "RtApiWasapi::wasapiThread: Unable to instantiate stream abort thread."; + goto Exit; + } + else if ( !CloseHandle( threadHandle ) ) { + errorType = RtAudioError::THREAD_ERROR; + errorText = "RtApiWasapi::wasapiThread: Unable to close stream abort thread handle."; + goto Exit; + } + + callbackStopped = true; + } + } + } + + // Callback Output + // =============== + // 1. Convert callback buffer to stream format + // 2. Convert callback buffer to stream sample rate and channel count + // 3. Push callback buffer into outputBuffer + + if ( renderAudioClient && callbackPulled ) + { + // if the last call to renderBuffer.PushBuffer() was successful + if ( callbackPushed || convBufferSize == 0 ) + { + if ( stream_.doConvertBuffer[OUTPUT] ) + { + // Convert callback buffer to stream format + convertBuffer( stream_.deviceBuffer, + stream_.userBuffer[OUTPUT], + stream_.convertInfo[OUTPUT] ); + + } + 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(); + + // update stream state + stream_.state = STREAM_STOPPED; + + if ( !errorText.empty() ) + { + errorText_ = errorText; + 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 <windows.h> +#include <process.h> +#include <mmsystem.h> +#include <mmreg.h> +#include <dsound.h> +#include <assert.h> +#include <algorithm> + +#if defined(__MINGW32__) + // missing from latest mingw winapi +#define WAVE_FORMAT_96M08 0x00010000 /* 96 kHz, Mono, 8-bit */ +#define WAVE_FORMAT_96S08 0x00020000 /* 96 kHz, Stereo, 8-bit */ +#define WAVE_FORMAT_96M16 0x00040000 /* 96 kHz, Mono, 16-bit */ +#define WAVE_FORMAT_96S16 0x00080000 /* 96 kHz, Stereo, 16-bit */ +#endif + +#define MINIMUM_DEVICE_BUFFER_SIZE 32768 + +#ifdef _MSC_VER // if Microsoft Visual C++ +#pragma comment( lib, "winmm.lib" ) // then, auto-link winmm.lib. Otherwise, it has to be added manually. +#endif + +static inline DWORD dsPointerBetween( DWORD pointer, DWORD laterPointer, DWORD earlierPointer, DWORD bufferSize ) +{ + if ( pointer > bufferSize ) pointer -= bufferSize; + if ( laterPointer < earlierPointer ) laterPointer += bufferSize; + if ( pointer < earlierPointer ) pointer += bufferSize; + return pointer >= earlierPointer && pointer < laterPointer; +} + +// A structure to hold various information related to the DirectSound +// API implementation. +struct DsHandle { + unsigned int drainCounter; // Tracks callback counts when draining + bool internalDrain; // Indicates if stop is initiated from callback or not. + void *id[2]; + void *buffer[2]; + bool xrun[2]; + UINT bufferPointer[2]; + DWORD dsBufferSize[2]; + DWORD dsPointerLeadTime[2]; // the number of bytes ahead of the safe pointer to lead by. + HANDLE condition; + + DsHandle() + :drainCounter(0), internalDrain(false) { id[0] = 0; id[1] = 0; buffer[0] = 0; buffer[1] = 0; xrun[0] = false; xrun[1] = false; bufferPointer[0] = 0; bufferPointer[1] = 0; } +}; + +// Declarations for utility functions, callbacks, and structures +// specific to the DirectSound implementation. +static BOOL CALLBACK deviceQueryCallback( LPGUID lpguid, + LPCTSTR description, + LPCTSTR module, + LPVOID lpContext ); + +static const char* getErrorString( int code ); + +static unsigned __stdcall callbackHandler( void *ptr ); + +struct DsDevice { + LPGUID id[2]; + bool validId[2]; + bool found; + std::string name; + + DsDevice() + : found(false) { validId[0] = false; validId[1] = false; } +}; + +struct DsProbeData { + bool isInput; + std::vector<struct DsDevice>* dsDevices; +}; + +RtApiDs :: RtApiDs() +{ + // Dsound will run both-threaded. If CoInitialize fails, then just + // accept whatever the mainline chose for a threading model. + coInitialized_ = false; + HRESULT hr = CoInitialize( NULL ); + if ( !FAILED( hr ) ) coInitialized_ = true; +} + +RtApiDs :: ~RtApiDs() +{ + if ( stream_.state != STREAM_CLOSED ) closeStream(); + if ( coInitialized_ ) CoUninitialize(); // balanced call. +} + +// The DirectSound default output is always the first device. +unsigned int RtApiDs :: getDefaultOutputDevice( void ) +{ + return 0; +} + +// The DirectSound default input is always the first input device, +// which is the first capture device enumerated. +unsigned int RtApiDs :: getDefaultInputDevice( void ) +{ + return 0; +} + +unsigned int RtApiDs :: getDeviceCount( void ) +{ + // Set query flag for previously found devices to false, so that we + // can check for any devices that have disappeared. + for ( unsigned int i=0; i<dsDevices.size(); i++ ) + dsDevices[i].found = false; + + // Query DirectSound devices. + struct DsProbeData probeInfo; + probeInfo.isInput = false; + probeInfo.dsDevices = &dsDevices; + HRESULT result = DirectSoundEnumerate( (LPDSENUMCALLBACK) deviceQueryCallback, &probeInfo ); + if ( FAILED( result ) ) { + errorStream_ << "RtApiDs::getDeviceCount: error (" << getErrorString( result ) << ") enumerating output devices!"; + errorText_ = errorStream_.str(); + error( RtAudioError::WARNING ); + } + + // Query DirectSoundCapture devices. + probeInfo.isInput = true; + result = DirectSoundCaptureEnumerate( (LPDSENUMCALLBACK) deviceQueryCallback, &probeInfo ); + if ( FAILED( result ) ) { + errorStream_ << "RtApiDs::getDeviceCount: error (" << getErrorString( result ) << ") enumerating input devices!"; + errorText_ = errorStream_.str(); + error( RtAudioError::WARNING ); + } + + // Clean out any devices that may have disappeared (code update submitted by Eli Zehngut). + for ( unsigned int i=0; i<dsDevices.size(); ) { + if ( dsDevices[i].found == false ) dsDevices.erase( dsDevices.begin() + i ); + else i++; + } + + return static_cast<unsigned int>(dsDevices.size()); +} + +RtAudio::DeviceInfo RtApiDs :: getDeviceInfo( unsigned int device ) +{ + RtAudio::DeviceInfo info; + info.probed = false; + + if ( dsDevices.size() == 0 ) { + // Force a query of all devices + getDeviceCount(); + if ( dsDevices.size() == 0 ) { + errorText_ = "RtApiDs::getDeviceInfo: no devices found!"; + error( RtAudioError::INVALID_USE ); + return info; + } + } + + if ( device >= dsDevices.size() ) { + errorText_ = "RtApiDs::getDeviceInfo: device ID is invalid!"; + error( RtAudioError::INVALID_USE ); + return info; + } + + HRESULT result; + if ( dsDevices[ device ].validId[0] == false ) goto probeInput; + + LPDIRECTSOUND output; + DSCAPS outCaps; + result = DirectSoundCreate( dsDevices[ device ].id[0], &output, NULL ); + if ( FAILED( result ) ) { + errorStream_ << "RtApiDs::getDeviceInfo: error (" << getErrorString( result ) << ") opening output device (" << dsDevices[ device ].name << ")!"; + errorText_ = errorStream_.str(); + error( RtAudioError::WARNING ); + goto probeInput; + } + + outCaps.dwSize = sizeof( outCaps ); + result = output->GetCaps( &outCaps ); + if ( FAILED( result ) ) { + output->Release(); + errorStream_ << "RtApiDs::getDeviceInfo: error (" << getErrorString( result ) << ") getting capabilities!"; + errorText_ = errorStream_.str(); + error( RtAudioError::WARNING ); + goto probeInput; + } + + // Get output channel information. + info.outputChannels = ( outCaps.dwFlags & DSCAPS_PRIMARYSTEREO ) ? 2 : 1; + + // Get sample rate information. + info.sampleRates.clear(); + for ( unsigned int k=0; k<MAX_SAMPLE_RATES; k++ ) { + if ( SAMPLE_RATES[k] >= (unsigned int) outCaps.dwMinSecondarySampleRate && + SAMPLE_RATES[k] <= (unsigned int) outCaps.dwMaxSecondarySampleRate ) { + info.sampleRates.push_back( SAMPLE_RATES[k] ); + + if ( !info.preferredSampleRate || ( SAMPLE_RATES[k] <= 48000 && SAMPLE_RATES[k] > info.preferredSampleRate ) ) + info.preferredSampleRate = SAMPLE_RATES[k]; + } + } + + // Get format information. + if ( outCaps.dwFlags & DSCAPS_PRIMARY16BIT ) info.nativeFormats |= RTAUDIO_SINT16; + if ( outCaps.dwFlags & DSCAPS_PRIMARY8BIT ) info.nativeFormats |= RTAUDIO_SINT8; + + output->Release(); + + if ( getDefaultOutputDevice() == device ) + info.isDefaultOutput = true; + + if ( dsDevices[ device ].validId[1] == false ) { + info.name = dsDevices[ device ].name; + info.probed = true; + return info; + } + + probeInput: + + LPDIRECTSOUNDCAPTURE input; + result = DirectSoundCaptureCreate( dsDevices[ device ].id[1], &input, NULL ); + if ( FAILED( result ) ) { + errorStream_ << "RtApiDs::getDeviceInfo: error (" << getErrorString( result ) << ") opening input device (" << dsDevices[ device ].name << ")!"; + errorText_ = errorStream_.str(); + error( RtAudioError::WARNING ); + return info; + } + + DSCCAPS inCaps; + inCaps.dwSize = sizeof( inCaps ); + result = input->GetCaps( &inCaps ); + if ( FAILED( result ) ) { + input->Release(); + errorStream_ << "RtApiDs::getDeviceInfo: error (" << getErrorString( result ) << ") getting object capabilities (" << dsDevices[ device ].name << ")!"; + errorText_ = errorStream_.str(); + error( RtAudioError::WARNING ); + return info; + } + + // Get input channel information. + info.inputChannels = inCaps.dwChannels; + + // Get sample rate and format information. + std::vector<unsigned int> rates; + if ( inCaps.dwChannels >= 2 ) { + if ( inCaps.dwFormats & WAVE_FORMAT_1S16 ) info.nativeFormats |= RTAUDIO_SINT16; + if ( inCaps.dwFormats & WAVE_FORMAT_2S16 ) info.nativeFormats |= RTAUDIO_SINT16; + if ( inCaps.dwFormats & WAVE_FORMAT_4S16 ) info.nativeFormats |= RTAUDIO_SINT16; + if ( inCaps.dwFormats & WAVE_FORMAT_96S16 ) info.nativeFormats |= RTAUDIO_SINT16; + if ( inCaps.dwFormats & WAVE_FORMAT_1S08 ) info.nativeFormats |= RTAUDIO_SINT8; + if ( inCaps.dwFormats & WAVE_FORMAT_2S08 ) info.nativeFormats |= RTAUDIO_SINT8; + if ( inCaps.dwFormats & WAVE_FORMAT_4S08 ) info.nativeFormats |= RTAUDIO_SINT8; + if ( inCaps.dwFormats & WAVE_FORMAT_96S08 ) info.nativeFormats |= RTAUDIO_SINT8; + + if ( info.nativeFormats & RTAUDIO_SINT16 ) { + if ( inCaps.dwFormats & WAVE_FORMAT_1S16 ) rates.push_back( 11025 ); + if ( inCaps.dwFormats & WAVE_FORMAT_2S16 ) rates.push_back( 22050 ); + if ( inCaps.dwFormats & WAVE_FORMAT_4S16 ) rates.push_back( 44100 ); + if ( inCaps.dwFormats & WAVE_FORMAT_96S16 ) rates.push_back( 96000 ); + } + else if ( info.nativeFormats & RTAUDIO_SINT8 ) { + if ( inCaps.dwFormats & WAVE_FORMAT_1S08 ) rates.push_back( 11025 ); + if ( inCaps.dwFormats & WAVE_FORMAT_2S08 ) rates.push_back( 22050 ); + if ( inCaps.dwFormats & WAVE_FORMAT_4S08 ) rates.push_back( 44100 ); + if ( inCaps.dwFormats & WAVE_FORMAT_96S08 ) rates.push_back( 96000 ); + } + } + else if ( inCaps.dwChannels == 1 ) { + if ( inCaps.dwFormats & WAVE_FORMAT_1M16 ) info.nativeFormats |= RTAUDIO_SINT16; + if ( inCaps.dwFormats & WAVE_FORMAT_2M16 ) info.nativeFormats |= RTAUDIO_SINT16; + if ( inCaps.dwFormats & WAVE_FORMAT_4M16 ) info.nativeFormats |= RTAUDIO_SINT16; + if ( inCaps.dwFormats & WAVE_FORMAT_96M16 ) info.nativeFormats |= RTAUDIO_SINT16; + if ( inCaps.dwFormats & WAVE_FORMAT_1M08 ) info.nativeFormats |= RTAUDIO_SINT8; + if ( inCaps.dwFormats & WAVE_FORMAT_2M08 ) info.nativeFormats |= RTAUDIO_SINT8; + if ( inCaps.dwFormats & WAVE_FORMAT_4M08 ) info.nativeFormats |= RTAUDIO_SINT8; + if ( inCaps.dwFormats & WAVE_FORMAT_96M08 ) info.nativeFormats |= RTAUDIO_SINT8; + + if ( info.nativeFormats & RTAUDIO_SINT16 ) { + if ( inCaps.dwFormats & WAVE_FORMAT_1M16 ) rates.push_back( 11025 ); + if ( inCaps.dwFormats & WAVE_FORMAT_2M16 ) rates.push_back( 22050 ); + if ( inCaps.dwFormats & WAVE_FORMAT_4M16 ) rates.push_back( 44100 ); + if ( inCaps.dwFormats & WAVE_FORMAT_96M16 ) rates.push_back( 96000 ); + } + else if ( info.nativeFormats & RTAUDIO_SINT8 ) { + if ( inCaps.dwFormats & WAVE_FORMAT_1M08 ) rates.push_back( 11025 ); + if ( inCaps.dwFormats & WAVE_FORMAT_2M08 ) rates.push_back( 22050 ); + if ( inCaps.dwFormats & WAVE_FORMAT_4M08 ) rates.push_back( 44100 ); + if ( inCaps.dwFormats & WAVE_FORMAT_96M08 ) rates.push_back( 96000 ); + } + } + else info.inputChannels = 0; // technically, this would be an error + + input->Release(); + + if ( info.inputChannels == 0 ) return info; + + // Copy the supported rates to the info structure but avoid duplication. + bool found; + for ( unsigned int i=0; i<rates.size(); i++ ) { + found = false; + for ( unsigned int j=0; j<info.sampleRates.size(); j++ ) { + if ( rates[i] == info.sampleRates[j] ) { + found = true; + break; + } + } + if ( found == false ) info.sampleRates.push_back( rates[i] ); + } + std::sort( info.sampleRates.begin(), info.sampleRates.end() ); + + // If device opens for both playback and capture, we determine the channels. + if ( info.outputChannels > 0 && info.inputChannels > 0 ) + info.duplexChannels = (info.outputChannels > info.inputChannels) ? info.inputChannels : info.outputChannels; + + if ( device == 0 ) info.isDefaultInput = true; + + // Copy name and return. + info.name = dsDevices[ device ].name; + info.probed = true; + return info; +} + +bool RtApiDs :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels, + unsigned int firstChannel, unsigned int sampleRate, + RtAudioFormat format, unsigned int *bufferSize, + RtAudio::StreamOptions *options ) +{ + if ( channels + firstChannel > 2 ) { + errorText_ = "RtApiDs::probeDeviceOpen: DirectSound does not support more than 2 channels per device."; + return FAILURE; + } + + size_t nDevices = dsDevices.size(); + if ( nDevices == 0 ) { + // This should not happen because a check is made before this function is called. + errorText_ = "RtApiDs::probeDeviceOpen: no devices found!"; + return FAILURE; + } + + if ( device >= nDevices ) { + // This should not happen because a check is made before this function is called. + errorText_ = "RtApiDs::probeDeviceOpen: device ID is invalid!"; + return FAILURE; + } + + if ( mode == OUTPUT ) { + if ( dsDevices[ device ].validId[0] == false ) { + errorStream_ << "RtApiDs::probeDeviceOpen: device (" << device << ") does not support output!"; + errorText_ = errorStream_.str(); + return FAILURE; + } + } + else { // mode == INPUT + if ( dsDevices[ device ].validId[1] == false ) { + errorStream_ << "RtApiDs::probeDeviceOpen: device (" << device << ") does not support input!"; + errorText_ = errorStream_.str(); + return FAILURE; + } + } + + // According to a note in PortAudio, using GetDesktopWindow() + // instead of GetForegroundWindow() is supposed to avoid problems + // that occur when the application's window is not the foreground + // window. Also, if the application window closes before the + // DirectSound buffer, DirectSound can crash. In the past, I had + // problems when using GetDesktopWindow() but it seems fine now + // (January 2010). I'll leave it commented here. + // HWND hWnd = GetForegroundWindow(); + HWND hWnd = GetDesktopWindow(); + + // Check the numberOfBuffers parameter and limit the lowest value to + // two. This is a judgement call and a value of two is probably too + // low for capture, but it should work for playback. + int nBuffers = 0; + if ( options ) nBuffers = options->numberOfBuffers; + if ( options && options->flags & RTAUDIO_MINIMIZE_LATENCY ) nBuffers = 2; + if ( nBuffers < 2 ) nBuffers = 3; + + // Check the lower range of the user-specified buffer size and set + // (arbitrarily) to a lower bound of 32. + if ( *bufferSize < 32 ) *bufferSize = 32; + + // Create the wave format structure. The data format setting will + // be determined later. + WAVEFORMATEX waveFormat; + ZeroMemory( &waveFormat, sizeof(WAVEFORMATEX) ); + waveFormat.wFormatTag = WAVE_FORMAT_PCM; + waveFormat.nChannels = channels + firstChannel; + waveFormat.nSamplesPerSec = (unsigned long) sampleRate; + + // Determine the device buffer size. By default, we'll use the value + // defined above (32K), but we will grow it to make allowances for + // very large software buffer sizes. + DWORD dsBufferSize = MINIMUM_DEVICE_BUFFER_SIZE; + DWORD dsPointerLeadTime = 0; + + void *ohandle = 0, *bhandle = 0; + HRESULT result; + if ( mode == OUTPUT ) { + + LPDIRECTSOUND output; + result = DirectSoundCreate( dsDevices[ device ].id[0], &output, NULL ); + if ( FAILED( result ) ) { + errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") opening output device (" << dsDevices[ device ].name << ")!"; + errorText_ = errorStream_.str(); + return FAILURE; + } + + DSCAPS outCaps; + outCaps.dwSize = sizeof( outCaps ); + result = output->GetCaps( &outCaps ); + if ( FAILED( result ) ) { + output->Release(); + errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") getting capabilities (" << dsDevices[ device ].name << ")!"; + errorText_ = errorStream_.str(); + return FAILURE; + } + + // Check channel information. + if ( channels + firstChannel == 2 && !( outCaps.dwFlags & DSCAPS_PRIMARYSTEREO ) ) { + errorStream_ << "RtApiDs::getDeviceInfo: the output device (" << dsDevices[ device ].name << ") does not support stereo playback."; + errorText_ = errorStream_.str(); + return FAILURE; + } + + // Check format information. Use 16-bit format unless not + // supported or user requests 8-bit. + if ( outCaps.dwFlags & DSCAPS_PRIMARY16BIT && + !( format == RTAUDIO_SINT8 && outCaps.dwFlags & DSCAPS_PRIMARY8BIT ) ) { + waveFormat.wBitsPerSample = 16; + stream_.deviceFormat[mode] = RTAUDIO_SINT16; + } + else { + waveFormat.wBitsPerSample = 8; + stream_.deviceFormat[mode] = RTAUDIO_SINT8; + } + stream_.userFormat = format; + + // Update wave format structure and buffer information. + waveFormat.nBlockAlign = waveFormat.nChannels * waveFormat.wBitsPerSample / 8; + waveFormat.nAvgBytesPerSec = waveFormat.nSamplesPerSec * waveFormat.nBlockAlign; + dsPointerLeadTime = nBuffers * (*bufferSize) * (waveFormat.wBitsPerSample / 8) * channels; + + // If the user wants an even bigger buffer, increase the device buffer size accordingly. + while ( dsPointerLeadTime * 2U > dsBufferSize ) + dsBufferSize *= 2; + + // Set cooperative level to DSSCL_EXCLUSIVE ... sound stops when window focus changes. + // result = output->SetCooperativeLevel( hWnd, DSSCL_EXCLUSIVE ); + // Set cooperative level to DSSCL_PRIORITY ... sound remains when window focus changes. + result = output->SetCooperativeLevel( hWnd, DSSCL_PRIORITY ); + if ( FAILED( result ) ) { + output->Release(); + errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") setting cooperative level (" << dsDevices[ device ].name << ")!"; + errorText_ = errorStream_.str(); + return FAILURE; + } + + // Even though we will write to the secondary buffer, we need to + // access the primary buffer to set the correct output format + // (since the default is 8-bit, 22 kHz!). Setup the DS primary + // buffer description. + DSBUFFERDESC bufferDescription; + ZeroMemory( &bufferDescription, sizeof( DSBUFFERDESC ) ); + bufferDescription.dwSize = sizeof( DSBUFFERDESC ); + bufferDescription.dwFlags = DSBCAPS_PRIMARYBUFFER; + + // Obtain the primary buffer + LPDIRECTSOUNDBUFFER buffer; + result = output->CreateSoundBuffer( &bufferDescription, &buffer, NULL ); + if ( FAILED( result ) ) { + output->Release(); + errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") accessing primary buffer (" << dsDevices[ device ].name << ")!"; + errorText_ = errorStream_.str(); + return FAILURE; + } + + // Set the primary DS buffer sound format. + result = buffer->SetFormat( &waveFormat ); + if ( FAILED( result ) ) { + output->Release(); + errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") setting primary buffer format (" << dsDevices[ device ].name << ")!"; + errorText_ = errorStream_.str(); + return FAILURE; + } + + // Setup the secondary DS buffer description. + ZeroMemory( &bufferDescription, sizeof( DSBUFFERDESC ) ); + bufferDescription.dwSize = sizeof( DSBUFFERDESC ); + bufferDescription.dwFlags = ( DSBCAPS_STICKYFOCUS | + DSBCAPS_GLOBALFOCUS | + DSBCAPS_GETCURRENTPOSITION2 | + DSBCAPS_LOCHARDWARE ); // Force hardware mixing + bufferDescription.dwBufferBytes = dsBufferSize; + bufferDescription.lpwfxFormat = &waveFormat; + + // Try to create the secondary DS buffer. If that doesn't work, + // try to use software mixing. Otherwise, there's a problem. + result = output->CreateSoundBuffer( &bufferDescription, &buffer, NULL ); + if ( FAILED( result ) ) { + bufferDescription.dwFlags = ( DSBCAPS_STICKYFOCUS | + DSBCAPS_GLOBALFOCUS | + DSBCAPS_GETCURRENTPOSITION2 | + DSBCAPS_LOCSOFTWARE ); // Force software mixing + result = output->CreateSoundBuffer( &bufferDescription, &buffer, NULL ); + if ( FAILED( result ) ) { + output->Release(); + errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") creating secondary buffer (" << dsDevices[ device ].name << ")!"; + errorText_ = errorStream_.str(); + return FAILURE; + } + } + + // Get the buffer size ... might be different from what we specified. + DSBCAPS dsbcaps; + dsbcaps.dwSize = sizeof( DSBCAPS ); + result = buffer->GetCaps( &dsbcaps ); + if ( FAILED( result ) ) { + output->Release(); + buffer->Release(); + errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") getting buffer settings (" << dsDevices[ device ].name << ")!"; + errorText_ = errorStream_.str(); + return FAILURE; + } + + dsBufferSize = dsbcaps.dwBufferBytes; + + // Lock the DS buffer + LPVOID audioPtr; + DWORD dataLen; + result = buffer->Lock( 0, dsBufferSize, &audioPtr, &dataLen, NULL, NULL, 0 ); + if ( FAILED( result ) ) { + output->Release(); + buffer->Release(); + errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") locking buffer (" << dsDevices[ device ].name << ")!"; + errorText_ = errorStream_.str(); + return FAILURE; + } + + // Zero the DS buffer + ZeroMemory( audioPtr, dataLen ); + + // Unlock the DS buffer + result = buffer->Unlock( audioPtr, dataLen, NULL, 0 ); + if ( FAILED( result ) ) { + output->Release(); + buffer->Release(); + errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") unlocking buffer (" << dsDevices[ device ].name << ")!"; + errorText_ = errorStream_.str(); + return FAILURE; + } + + ohandle = (void *) output; + bhandle = (void *) buffer; + } + + if ( mode == INPUT ) { + + LPDIRECTSOUNDCAPTURE input; + result = DirectSoundCaptureCreate( dsDevices[ device ].id[1], &input, NULL ); + if ( FAILED( result ) ) { + errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") opening input device (" << dsDevices[ device ].name << ")!"; + errorText_ = errorStream_.str(); + return FAILURE; + } + + DSCCAPS inCaps; + inCaps.dwSize = sizeof( inCaps ); + result = input->GetCaps( &inCaps ); + if ( FAILED( result ) ) { + input->Release(); + errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") getting input capabilities (" << dsDevices[ device ].name << ")!"; + errorText_ = errorStream_.str(); + return FAILURE; + } + + // Check channel information. + if ( inCaps.dwChannels < channels + firstChannel ) { + errorText_ = "RtApiDs::getDeviceInfo: the input device does not support requested input channels."; + return FAILURE; + } + + // Check format information. Use 16-bit format unless user + // requests 8-bit. + DWORD deviceFormats; + if ( channels + firstChannel == 2 ) { + deviceFormats = WAVE_FORMAT_1S08 | WAVE_FORMAT_2S08 | WAVE_FORMAT_4S08 | WAVE_FORMAT_96S08; + if ( format == RTAUDIO_SINT8 && inCaps.dwFormats & deviceFormats ) { + waveFormat.wBitsPerSample = 8; + stream_.deviceFormat[mode] = RTAUDIO_SINT8; + } + else { // assume 16-bit is supported + waveFormat.wBitsPerSample = 16; + stream_.deviceFormat[mode] = RTAUDIO_SINT16; + } + } + else { // channel == 1 + deviceFormats = WAVE_FORMAT_1M08 | WAVE_FORMAT_2M08 | WAVE_FORMAT_4M08 | WAVE_FORMAT_96M08; + if ( format == RTAUDIO_SINT8 && inCaps.dwFormats & deviceFormats ) { + waveFormat.wBitsPerSample = 8; + stream_.deviceFormat[mode] = RTAUDIO_SINT8; + } + else { // assume 16-bit is supported + waveFormat.wBitsPerSample = 16; + stream_.deviceFormat[mode] = RTAUDIO_SINT16; + } + } + stream_.userFormat = format; + + // Update wave format structure and buffer information. + waveFormat.nBlockAlign = waveFormat.nChannels * waveFormat.wBitsPerSample / 8; + waveFormat.nAvgBytesPerSec = waveFormat.nSamplesPerSec * waveFormat.nBlockAlign; + dsPointerLeadTime = nBuffers * (*bufferSize) * (waveFormat.wBitsPerSample / 8) * channels; + + // If the user wants an even bigger buffer, increase the device buffer size accordingly. + while ( dsPointerLeadTime * 2U > dsBufferSize ) + dsBufferSize *= 2; + + // Setup the secondary DS buffer description. + DSCBUFFERDESC bufferDescription; + ZeroMemory( &bufferDescription, sizeof( DSCBUFFERDESC ) ); + bufferDescription.dwSize = sizeof( DSCBUFFERDESC ); + bufferDescription.dwFlags = 0; + bufferDescription.dwReserved = 0; + bufferDescription.dwBufferBytes = dsBufferSize; + bufferDescription.lpwfxFormat = &waveFormat; + + // Create the capture buffer. + LPDIRECTSOUNDCAPTUREBUFFER buffer; + result = input->CreateCaptureBuffer( &bufferDescription, &buffer, NULL ); + if ( FAILED( result ) ) { + input->Release(); + errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") creating input buffer (" << dsDevices[ device ].name << ")!"; + errorText_ = errorStream_.str(); + return FAILURE; + } + + // Get the buffer size ... might be different from what we specified. + DSCBCAPS dscbcaps; + dscbcaps.dwSize = sizeof( DSCBCAPS ); + result = buffer->GetCaps( &dscbcaps ); + if ( FAILED( result ) ) { + input->Release(); + buffer->Release(); + errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") getting buffer settings (" << dsDevices[ device ].name << ")!"; + errorText_ = errorStream_.str(); + return FAILURE; + } + + dsBufferSize = dscbcaps.dwBufferBytes; + + // NOTE: We could have a problem here if this is a duplex stream + // and the play and capture hardware buffer sizes are different + // (I'm actually not sure if that is a problem or not). + // Currently, we are not verifying that. + + // Lock the capture buffer + LPVOID audioPtr; + DWORD dataLen; + result = buffer->Lock( 0, dsBufferSize, &audioPtr, &dataLen, NULL, NULL, 0 ); + if ( FAILED( result ) ) { + input->Release(); + buffer->Release(); + errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") locking input buffer (" << dsDevices[ device ].name << ")!"; + errorText_ = errorStream_.str(); + return FAILURE; + } + + // Zero the buffer + ZeroMemory( audioPtr, dataLen ); + + // Unlock the buffer + result = buffer->Unlock( audioPtr, dataLen, NULL, 0 ); + if ( FAILED( result ) ) { + input->Release(); + buffer->Release(); + errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") unlocking input buffer (" << dsDevices[ device ].name << ")!"; + errorText_ = errorStream_.str(); + return FAILURE; + } + + ohandle = (void *) input; + bhandle = (void *) buffer; + } + + // Set various stream parameters + DsHandle *handle = 0; + stream_.nDeviceChannels[mode] = channels + firstChannel; + stream_.nUserChannels[mode] = channels; + stream_.bufferSize = *bufferSize; + stream_.channelOffset[mode] = firstChannel; + stream_.deviceInterleaved[mode] = true; + if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) stream_.userInterleaved = false; + else stream_.userInterleaved = true; + + // Set flag for buffer conversion + stream_.doConvertBuffer[mode] = false; + if (stream_.nUserChannels[mode] != stream_.nDeviceChannels[mode]) + stream_.doConvertBuffer[mode] = true; + if (stream_.userFormat != stream_.deviceFormat[mode]) + stream_.doConvertBuffer[mode] = true; + if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] && + stream_.nUserChannels[mode] > 1 ) + stream_.doConvertBuffer[mode] = true; + + // Allocate necessary internal buffers + long bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat ); + stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 ); + if ( stream_.userBuffer[mode] == NULL ) { + errorText_ = "RtApiDs::probeDeviceOpen: error allocating user buffer memory."; + goto error; + } + + if ( stream_.doConvertBuffer[mode] ) { + + bool makeBuffer = true; + bufferBytes = stream_.nDeviceChannels[mode] * formatBytes( stream_.deviceFormat[mode] ); + if ( mode == INPUT ) { + if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) { + unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] ); + if ( bufferBytes <= (long) bytesOut ) makeBuffer = false; + } + } + + if ( makeBuffer ) { + bufferBytes *= *bufferSize; + if ( stream_.deviceBuffer ) free( stream_.deviceBuffer ); + stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 ); + if ( stream_.deviceBuffer == NULL ) { + errorText_ = "RtApiDs::probeDeviceOpen: error allocating device buffer memory."; + goto error; + } + } + } + + // Allocate our DsHandle structures for the stream. + if ( stream_.apiHandle == 0 ) { + try { + handle = new DsHandle; + } + catch ( std::bad_alloc& ) { + errorText_ = "RtApiDs::probeDeviceOpen: error allocating AsioHandle memory."; + goto error; + } + + // Create a manual-reset event. + handle->condition = CreateEvent( NULL, // no security + TRUE, // manual-reset + FALSE, // non-signaled initially + NULL ); // unnamed + stream_.apiHandle = (void *) handle; + } + else + handle = (DsHandle *) stream_.apiHandle; + handle->id[mode] = ohandle; + handle->buffer[mode] = bhandle; + handle->dsBufferSize[mode] = dsBufferSize; + handle->dsPointerLeadTime[mode] = dsPointerLeadTime; + + stream_.device[mode] = device; + stream_.state = STREAM_STOPPED; + if ( stream_.mode == OUTPUT && mode == INPUT ) + // We had already set up an output stream. + stream_.mode = DUPLEX; + else + stream_.mode = mode; + stream_.nBuffers = nBuffers; + stream_.sampleRate = sampleRate; + + // Setup the buffer conversion information structure. + if ( stream_.doConvertBuffer[mode] ) setConvertInfo( mode, firstChannel ); + + // Setup the callback thread. + if ( stream_.callbackInfo.isRunning == false ) { + unsigned threadId; + stream_.callbackInfo.isRunning = true; + stream_.callbackInfo.object = (void *) this; + stream_.callbackInfo.thread = _beginthreadex( NULL, 0, &callbackHandler, + &stream_.callbackInfo, 0, &threadId ); + if ( stream_.callbackInfo.thread == 0 ) { + errorText_ = "RtApiDs::probeDeviceOpen: error creating callback thread!"; + goto error; + } + + // Boost DS thread priority + SetThreadPriority( (HANDLE) stream_.callbackInfo.thread, THREAD_PRIORITY_HIGHEST ); + } + return SUCCESS; + + error: + if ( handle ) { + if ( handle->buffer[0] ) { // the object pointer can be NULL and valid + LPDIRECTSOUND object = (LPDIRECTSOUND) handle->id[0]; + LPDIRECTSOUNDBUFFER buffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0]; + if ( buffer ) buffer->Release(); + object->Release(); + } + if ( handle->buffer[1] ) { + LPDIRECTSOUNDCAPTURE object = (LPDIRECTSOUNDCAPTURE) handle->id[1]; + LPDIRECTSOUNDCAPTUREBUFFER buffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1]; + if ( buffer ) buffer->Release(); + object->Release(); + } + CloseHandle( handle->condition ); + delete handle; + stream_.apiHandle = 0; + } + + for ( int i=0; i<2; i++ ) { + if ( stream_.userBuffer[i] ) { + free( stream_.userBuffer[i] ); + stream_.userBuffer[i] = 0; + } + } + + if ( stream_.deviceBuffer ) { + free( stream_.deviceBuffer ); + stream_.deviceBuffer = 0; + } + + stream_.state = STREAM_CLOSED; + return FAILURE; +} + +void RtApiDs :: closeStream() +{ + if ( stream_.state == STREAM_CLOSED ) { + errorText_ = "RtApiDs::closeStream(): no open stream to close!"; + error( RtAudioError::WARNING ); + return; + } + + // Stop the callback thread. + stream_.callbackInfo.isRunning = false; + WaitForSingleObject( (HANDLE) stream_.callbackInfo.thread, INFINITE ); + CloseHandle( (HANDLE) stream_.callbackInfo.thread ); + + DsHandle *handle = (DsHandle *) stream_.apiHandle; + if ( handle ) { + if ( handle->buffer[0] ) { // the object pointer can be NULL and valid + LPDIRECTSOUND object = (LPDIRECTSOUND) handle->id[0]; + LPDIRECTSOUNDBUFFER buffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0]; + if ( buffer ) { + buffer->Stop(); + buffer->Release(); + } + object->Release(); + } + if ( handle->buffer[1] ) { + LPDIRECTSOUNDCAPTURE object = (LPDIRECTSOUNDCAPTURE) handle->id[1]; + LPDIRECTSOUNDCAPTUREBUFFER buffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1]; + if ( buffer ) { + buffer->Stop(); + buffer->Release(); + } + object->Release(); + } + CloseHandle( handle->condition ); + delete handle; + stream_.apiHandle = 0; + } + + for ( int i=0; i<2; i++ ) { + if ( stream_.userBuffer[i] ) { + free( stream_.userBuffer[i] ); + stream_.userBuffer[i] = 0; + } + } + + if ( stream_.deviceBuffer ) { + free( stream_.deviceBuffer ); + stream_.deviceBuffer = 0; + } + + stream_.mode = UNINITIALIZED; + stream_.state = STREAM_CLOSED; +} + +void RtApiDs :: startStream() +{ + verifyStream(); + if ( stream_.state == STREAM_RUNNING ) { + errorText_ = "RtApiDs::startStream(): the stream is already running!"; + error( RtAudioError::WARNING ); + return; + } + + #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( RtAudioError::SYSTEM_ERROR ); +} + +void RtApiDs :: stopStream() +{ + verifyStream(); + if ( stream_.state == STREAM_STOPPED ) { + errorText_ = "RtApiDs::stopStream(): the stream is already stopped!"; + error( RtAudioError::WARNING ); + return; + } + + HRESULT result = 0; + LPVOID audioPtr; + DWORD dataLen; + DsHandle *handle = (DsHandle *) stream_.apiHandle; + if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { + if ( handle->drainCounter == 0 ) { + handle->drainCounter = 2; + WaitForSingleObject( handle->condition, INFINITE ); // block until signaled + } + + stream_.state = STREAM_STOPPED; + + MUTEX_LOCK( &stream_.mutex ); + + // Stop the buffer and clear memory + LPDIRECTSOUNDBUFFER buffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0]; + result = buffer->Stop(); + if ( FAILED( result ) ) { + errorStream_ << "RtApiDs::stopStream: error (" << getErrorString( result ) << ") stopping output buffer!"; + errorText_ = errorStream_.str(); + goto unlock; + } + + // Lock the buffer and clear it so that if we start to play again, + // we won't have old data playing. + result = buffer->Lock( 0, handle->dsBufferSize[0], &audioPtr, &dataLen, NULL, NULL, 0 ); + if ( FAILED( result ) ) { + errorStream_ << "RtApiDs::stopStream: error (" << getErrorString( result ) << ") locking output buffer!"; + errorText_ = errorStream_.str(); + goto unlock; + } + + // Zero the DS buffer + ZeroMemory( audioPtr, dataLen ); + + // Unlock the DS buffer + result = buffer->Unlock( audioPtr, dataLen, NULL, 0 ); + if ( FAILED( result ) ) { + errorStream_ << "RtApiDs::stopStream: error (" << getErrorString( result ) << ") unlocking output buffer!"; + errorText_ = errorStream_.str(); + goto unlock; + } + + // If we start playing again, we must begin at beginning of buffer. + handle->bufferPointer[0] = 0; + } + + if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) { + LPDIRECTSOUNDCAPTUREBUFFER buffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1]; + audioPtr = NULL; + dataLen = 0; + + stream_.state = STREAM_STOPPED; + + if ( stream_.mode != DUPLEX ) + MUTEX_LOCK( &stream_.mutex ); + + result = buffer->Stop(); + if ( FAILED( result ) ) { + errorStream_ << "RtApiDs::stopStream: error (" << getErrorString( result ) << ") stopping input buffer!"; + errorText_ = errorStream_.str(); + goto unlock; + } + + // Lock the buffer and clear it so that if we start to play again, + // we won't have old data playing. + result = buffer->Lock( 0, handle->dsBufferSize[1], &audioPtr, &dataLen, NULL, NULL, 0 ); + if ( FAILED( result ) ) { + errorStream_ << "RtApiDs::stopStream: error (" << getErrorString( result ) << ") locking input buffer!"; + errorText_ = errorStream_.str(); + goto unlock; + } + + // Zero the DS buffer + ZeroMemory( audioPtr, dataLen ); + + // Unlock the DS buffer + result = buffer->Unlock( audioPtr, dataLen, NULL, 0 ); + if ( FAILED( result ) ) { + errorStream_ << "RtApiDs::stopStream: error (" << getErrorString( result ) << ") unlocking input buffer!"; + errorText_ = errorStream_.str(); + goto unlock; + } + + // If we start recording again, we must begin at beginning of buffer. + handle->bufferPointer[1] = 0; + } + + unlock: + timeEndPeriod( 1 ); // revert to normal scheduler frequency on lesser windows. + MUTEX_UNLOCK( &stream_.mutex ); + + if ( FAILED( result ) ) error( RtAudioError::SYSTEM_ERROR ); +} + +void RtApiDs :: abortStream() +{ + verifyStream(); + if ( stream_.state == STREAM_STOPPED ) { + errorText_ = "RtApiDs::abortStream(): the stream is already stopped!"; + error( RtAudioError::WARNING ); + return; + } + + DsHandle *handle = (DsHandle *) stream_.apiHandle; + handle->drainCounter = 2; + + stopStream(); +} + +void RtApiDs :: callbackEvent() +{ + if ( stream_.state == STREAM_STOPPED || stream_.state == STREAM_STOPPING ) { + Sleep( 50 ); // sleep 50 milliseconds + return; + } + + if ( stream_.state == STREAM_CLOSED ) { + errorText_ = "RtApiDs::callbackEvent(): the stream is closed ... this shouldn't happen!"; + error( RtAudioError::WARNING ); + return; + } + + CallbackInfo *info = (CallbackInfo *) &stream_.callbackInfo; + DsHandle *handle = (DsHandle *) stream_.apiHandle; + + // Check if we were draining the stream and signal is finished. + if ( handle->drainCounter > stream_.nBuffers + 2 ) { + + stream_.state = STREAM_STOPPING; + if ( handle->internalDrain == false ) + SetEvent( handle->condition ); + else + stopStream(); + return; + } + + // Invoke user callback to get fresh output data UNLESS we are + // draining stream. + if ( handle->drainCounter == 0 ) { + RtAudioCallback callback = (RtAudioCallback) info->callback; + double streamTime = getStreamTime(); + RtAudioStreamStatus status = 0; + if ( stream_.mode != INPUT && handle->xrun[0] == true ) { + status |= RTAUDIO_OUTPUT_UNDERFLOW; + handle->xrun[0] = false; + } + if ( stream_.mode != OUTPUT && handle->xrun[1] == true ) { + status |= RTAUDIO_INPUT_OVERFLOW; + handle->xrun[1] = false; + } + int cbReturnValue = callback( stream_.userBuffer[0], stream_.userBuffer[1], + stream_.bufferSize, streamTime, status, info->userData ); + if ( cbReturnValue == 2 ) { + stream_.state = STREAM_STOPPING; + handle->drainCounter = 2; + abortStream(); + return; + } + else if ( cbReturnValue == 1 ) { + handle->drainCounter = 1; + handle->internalDrain = true; + } + } + + HRESULT result; + DWORD currentWritePointer, safeWritePointer; + DWORD currentReadPointer, safeReadPointer; + UINT nextWritePointer; + + LPVOID buffer1 = NULL; + LPVOID buffer2 = NULL; + DWORD bufferSize1 = 0; + DWORD bufferSize2 = 0; + + char *buffer; + long bufferBytes; + + MUTEX_LOCK( &stream_.mutex ); + if ( stream_.state == STREAM_STOPPED ) { + MUTEX_UNLOCK( &stream_.mutex ); + return; + } + + if ( buffersRolling == false ) { + if ( stream_.mode == DUPLEX ) { + //assert( handle->dsBufferSize[0] == handle->dsBufferSize[1] ); + + // It takes a while for the devices to get rolling. As a result, + // there's no guarantee that the capture and write device pointers + // will move in lockstep. Wait here for both devices to start + // rolling, and then set our buffer pointers accordingly. + // e.g. Crystal Drivers: the capture buffer starts up 5700 to 9600 + // bytes later than the write buffer. + + // Stub: a serious risk of having a pre-emptive scheduling round + // take place between the two GetCurrentPosition calls... but I'm + // really not sure how to solve the problem. Temporarily boost to + // Realtime priority, maybe; but I'm not sure what priority the + // DirectSound service threads run at. We *should* be roughly + // within a ms or so of correct. + + LPDIRECTSOUNDBUFFER dsWriteBuffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0]; + LPDIRECTSOUNDCAPTUREBUFFER dsCaptureBuffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1]; + + DWORD startSafeWritePointer, startSafeReadPointer; + + result = dsWriteBuffer->GetCurrentPosition( NULL, &startSafeWritePointer ); + if ( FAILED( result ) ) { + errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current write position!"; + errorText_ = errorStream_.str(); + MUTEX_UNLOCK( &stream_.mutex ); + error( RtAudioError::SYSTEM_ERROR ); + return; + } + result = dsCaptureBuffer->GetCurrentPosition( NULL, &startSafeReadPointer ); + if ( FAILED( result ) ) { + errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current read position!"; + errorText_ = errorStream_.str(); + MUTEX_UNLOCK( &stream_.mutex ); + error( RtAudioError::SYSTEM_ERROR ); + return; + } + while ( true ) { + result = dsWriteBuffer->GetCurrentPosition( NULL, &safeWritePointer ); + if ( FAILED( result ) ) { + errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current write position!"; + errorText_ = errorStream_.str(); + MUTEX_UNLOCK( &stream_.mutex ); + error( RtAudioError::SYSTEM_ERROR ); + return; + } + result = dsCaptureBuffer->GetCurrentPosition( NULL, &safeReadPointer ); + if ( FAILED( result ) ) { + errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current read position!"; + errorText_ = errorStream_.str(); + MUTEX_UNLOCK( &stream_.mutex ); + error( RtAudioError::SYSTEM_ERROR ); + return; + } + if ( safeWritePointer != startSafeWritePointer && safeReadPointer != startSafeReadPointer ) break; + Sleep( 1 ); + } + + //assert( handle->dsBufferSize[0] == handle->dsBufferSize[1] ); + + handle->bufferPointer[0] = safeWritePointer + handle->dsPointerLeadTime[0]; + if ( handle->bufferPointer[0] >= handle->dsBufferSize[0] ) handle->bufferPointer[0] -= handle->dsBufferSize[0]; + handle->bufferPointer[1] = safeReadPointer; + } + else if ( stream_.mode == OUTPUT ) { + + // Set the proper nextWritePosition after initial startup. + LPDIRECTSOUNDBUFFER dsWriteBuffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0]; + result = dsWriteBuffer->GetCurrentPosition( ¤tWritePointer, &safeWritePointer ); + if ( FAILED( result ) ) { + errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current write position!"; + errorText_ = errorStream_.str(); + MUTEX_UNLOCK( &stream_.mutex ); + error( RtAudioError::SYSTEM_ERROR ); + return; + } + handle->bufferPointer[0] = safeWritePointer + handle->dsPointerLeadTime[0]; + if ( handle->bufferPointer[0] >= handle->dsBufferSize[0] ) handle->bufferPointer[0] -= handle->dsBufferSize[0]; + } + + buffersRolling = true; + } + + if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { + + LPDIRECTSOUNDBUFFER dsBuffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0]; + + if ( handle->drainCounter > 1 ) { // write zeros to the output stream + bufferBytes = stream_.bufferSize * stream_.nUserChannels[0]; + bufferBytes *= formatBytes( stream_.userFormat ); + memset( stream_.userBuffer[0], 0, bufferBytes ); + } + + // Setup parameters and do buffer conversion if necessary. + if ( stream_.doConvertBuffer[0] ) { + buffer = stream_.deviceBuffer; + convertBuffer( buffer, stream_.userBuffer[0], stream_.convertInfo[0] ); + bufferBytes = stream_.bufferSize * stream_.nDeviceChannels[0]; + bufferBytes *= formatBytes( stream_.deviceFormat[0] ); + } + else { + buffer = stream_.userBuffer[0]; + bufferBytes = stream_.bufferSize * stream_.nUserChannels[0]; + bufferBytes *= formatBytes( stream_.userFormat ); + } + + // No byte swapping necessary in DirectSound implementation. + + // Ahhh ... windoze. 16-bit data is signed but 8-bit data is + // unsigned. So, we need to convert our signed 8-bit data here to + // unsigned. + if ( stream_.deviceFormat[0] == RTAUDIO_SINT8 ) + for ( int i=0; i<bufferBytes; i++ ) buffer[i] = (unsigned char) ( buffer[i] + 128 ); + + DWORD dsBufferSize = handle->dsBufferSize[0]; + nextWritePointer = handle->bufferPointer[0]; + + DWORD endWrite, leadPointer; + while ( true ) { + // Find out where the read and "safe write" pointers are. + result = dsBuffer->GetCurrentPosition( ¤tWritePointer, &safeWritePointer ); + if ( FAILED( result ) ) { + errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current write position!"; + errorText_ = errorStream_.str(); + MUTEX_UNLOCK( &stream_.mutex ); + error( RtAudioError::SYSTEM_ERROR ); + return; + } + + // We will copy our output buffer into the region between + // safeWritePointer and leadPointer. If leadPointer is not + // beyond the next endWrite position, wait until it is. + leadPointer = safeWritePointer + handle->dsPointerLeadTime[0]; + //std::cout << "safeWritePointer = " << safeWritePointer << ", leadPointer = " << leadPointer << ", nextWritePointer = " << nextWritePointer << std::endl; + if ( leadPointer > dsBufferSize ) leadPointer -= dsBufferSize; + if ( leadPointer < nextWritePointer ) leadPointer += dsBufferSize; // unwrap offset + endWrite = nextWritePointer + bufferBytes; + + // Check whether the entire write region is behind the play pointer. + if ( leadPointer >= endWrite ) break; + + // If we are here, then we must wait until the leadPointer advances + // beyond the end of our next write region. We use the + // Sleep() function to suspend operation until that happens. + double millis = ( endWrite - leadPointer ) * 1000.0; + millis /= ( formatBytes( stream_.deviceFormat[0]) * stream_.nDeviceChannels[0] * stream_.sampleRate); + if ( millis < 1.0 ) millis = 1.0; + Sleep( (DWORD) millis ); + } + + if ( dsPointerBetween( nextWritePointer, safeWritePointer, currentWritePointer, dsBufferSize ) + || dsPointerBetween( endWrite, safeWritePointer, currentWritePointer, dsBufferSize ) ) { + // We've strayed into the forbidden zone ... resync the read pointer. + handle->xrun[0] = true; + nextWritePointer = safeWritePointer + handle->dsPointerLeadTime[0] - bufferBytes; + if ( nextWritePointer >= dsBufferSize ) nextWritePointer -= dsBufferSize; + handle->bufferPointer[0] = nextWritePointer; + endWrite = nextWritePointer + bufferBytes; + } + + // Lock free space in the buffer + result = dsBuffer->Lock( nextWritePointer, bufferBytes, &buffer1, + &bufferSize1, &buffer2, &bufferSize2, 0 ); + if ( FAILED( result ) ) { + errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") locking buffer during playback!"; + errorText_ = errorStream_.str(); + MUTEX_UNLOCK( &stream_.mutex ); + error( RtAudioError::SYSTEM_ERROR ); + return; + } + + // Copy our buffer into the DS buffer + CopyMemory( buffer1, buffer, bufferSize1 ); + if ( buffer2 != NULL ) CopyMemory( buffer2, buffer+bufferSize1, bufferSize2 ); + + // Update our buffer offset and unlock sound buffer + dsBuffer->Unlock( buffer1, bufferSize1, buffer2, bufferSize2 ); + if ( FAILED( result ) ) { + errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") unlocking buffer during playback!"; + errorText_ = errorStream_.str(); + MUTEX_UNLOCK( &stream_.mutex ); + error( RtAudioError::SYSTEM_ERROR ); + return; + } + nextWritePointer = ( nextWritePointer + bufferSize1 + bufferSize2 ) % dsBufferSize; + handle->bufferPointer[0] = nextWritePointer; + } + + // Don't bother draining input + if ( handle->drainCounter ) { + handle->drainCounter++; + goto unlock; + } + + if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) { + + // Setup parameters. + if ( stream_.doConvertBuffer[1] ) { + buffer = stream_.deviceBuffer; + bufferBytes = stream_.bufferSize * stream_.nDeviceChannels[1]; + bufferBytes *= formatBytes( stream_.deviceFormat[1] ); + } + else { + buffer = stream_.userBuffer[1]; + bufferBytes = stream_.bufferSize * stream_.nUserChannels[1]; + bufferBytes *= formatBytes( stream_.userFormat ); + } + + LPDIRECTSOUNDCAPTUREBUFFER dsBuffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1]; + long nextReadPointer = handle->bufferPointer[1]; + DWORD dsBufferSize = handle->dsBufferSize[1]; + + // Find out where the write and "safe read" pointers are. + result = dsBuffer->GetCurrentPosition( ¤tReadPointer, &safeReadPointer ); + if ( FAILED( result ) ) { + errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current read position!"; + errorText_ = errorStream_.str(); + MUTEX_UNLOCK( &stream_.mutex ); + error( RtAudioError::SYSTEM_ERROR ); + return; + } + + if ( safeReadPointer < (DWORD)nextReadPointer ) safeReadPointer += dsBufferSize; // unwrap offset + DWORD endRead = nextReadPointer + bufferBytes; + + // Handling depends on whether we are INPUT or DUPLEX. + // If we're in INPUT mode then waiting is a good thing. If we're in DUPLEX mode, + // then a wait here will drag the write pointers into the forbidden zone. + // + // In DUPLEX mode, rather than wait, we will back off the read pointer until + // it's in a safe position. This causes dropouts, but it seems to be the only + // practical way to sync up the read and write pointers reliably, given the + // the very complex relationship between phase and increment of the read and write + // pointers. + // + // In order to minimize audible dropouts in DUPLEX mode, we will + // provide a pre-roll period of 0.5 seconds in which we return + // zeros from the read buffer while the pointers sync up. + + if ( stream_.mode == DUPLEX ) { + if ( safeReadPointer < endRead ) { + if ( duplexPrerollBytes <= 0 ) { + // Pre-roll time over. Be more agressive. + int adjustment = endRead-safeReadPointer; + + handle->xrun[1] = true; + // Two cases: + // - large adjustments: we've probably run out of CPU cycles, so just resync exactly, + // and perform fine adjustments later. + // - small adjustments: back off by twice as much. + if ( adjustment >= 2*bufferBytes ) + nextReadPointer = safeReadPointer-2*bufferBytes; + else + nextReadPointer = safeReadPointer-bufferBytes-adjustment; + + if ( nextReadPointer < 0 ) nextReadPointer += dsBufferSize; + + } + else { + // In pre=roll time. Just do it. + nextReadPointer = safeReadPointer - bufferBytes; + while ( nextReadPointer < 0 ) nextReadPointer += dsBufferSize; + } + endRead = nextReadPointer + bufferBytes; + } + } + else { // mode == INPUT + while ( safeReadPointer < endRead && stream_.callbackInfo.isRunning ) { + // See comments for playback. + double millis = (endRead - safeReadPointer) * 1000.0; + millis /= ( formatBytes(stream_.deviceFormat[1]) * stream_.nDeviceChannels[1] * stream_.sampleRate); + if ( millis < 1.0 ) millis = 1.0; + Sleep( (DWORD) millis ); + + // Wake up and find out where we are now. + result = dsBuffer->GetCurrentPosition( ¤tReadPointer, &safeReadPointer ); + if ( FAILED( result ) ) { + errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current read position!"; + errorText_ = errorStream_.str(); + MUTEX_UNLOCK( &stream_.mutex ); + error( RtAudioError::SYSTEM_ERROR ); + return; + } + + if ( safeReadPointer < (DWORD)nextReadPointer ) safeReadPointer += dsBufferSize; // unwrap offset + } + } + + // Lock free space in the buffer + result = dsBuffer->Lock( nextReadPointer, bufferBytes, &buffer1, + &bufferSize1, &buffer2, &bufferSize2, 0 ); + if ( FAILED( result ) ) { + errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") locking capture buffer!"; + errorText_ = errorStream_.str(); + MUTEX_UNLOCK( &stream_.mutex ); + error( RtAudioError::SYSTEM_ERROR ); + return; + } + + if ( duplexPrerollBytes <= 0 ) { + // Copy our buffer into the DS buffer + CopyMemory( buffer, buffer1, bufferSize1 ); + if ( buffer2 != NULL ) CopyMemory( buffer+bufferSize1, buffer2, bufferSize2 ); + } + else { + memset( buffer, 0, bufferSize1 ); + if ( buffer2 != NULL ) memset( buffer + bufferSize1, 0, bufferSize2 ); + duplexPrerollBytes -= bufferSize1 + bufferSize2; + } + + // Update our buffer offset and unlock sound buffer + nextReadPointer = ( nextReadPointer + bufferSize1 + bufferSize2 ) % dsBufferSize; + dsBuffer->Unlock( buffer1, bufferSize1, buffer2, bufferSize2 ); + if ( FAILED( result ) ) { + errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") unlocking capture buffer!"; + errorText_ = errorStream_.str(); + MUTEX_UNLOCK( &stream_.mutex ); + error( RtAudioError::SYSTEM_ERROR ); + return; + } + handle->bufferPointer[1] = nextReadPointer; + + // No byte swapping necessary in DirectSound implementation. + + // If necessary, convert 8-bit data from unsigned to signed. + if ( stream_.deviceFormat[1] == RTAUDIO_SINT8 ) + for ( int j=0; j<bufferBytes; j++ ) buffer[j] = (signed char) ( buffer[j] - 128 ); + + // Do buffer conversion if necessary. + if ( stream_.doConvertBuffer[1] ) + convertBuffer( stream_.userBuffer[1], stream_.deviceBuffer, stream_.convertInfo[1] ); + } + + unlock: + MUTEX_UNLOCK( &stream_.mutex ); + RtApi::tickStreamTime(); +} + +// Definitions for utility functions and callbacks +// specific to the DirectSound implementation. + +static unsigned __stdcall callbackHandler( void *ptr ) +{ + CallbackInfo *info = (CallbackInfo *) ptr; + RtApiDs *object = (RtApiDs *) info->object; + bool* isRunning = &info->isRunning; + + while ( *isRunning == true ) { + object->callbackEvent(); + } + + _endthreadex( 0 ); + return 0; +} + +static BOOL CALLBACK deviceQueryCallback( LPGUID lpguid, + LPCTSTR description, + LPCTSTR /*module*/, + LPVOID lpContext ) +{ + struct DsProbeData& probeInfo = *(struct DsProbeData*) lpContext; + std::vector<struct DsDevice>& dsDevices = *probeInfo.dsDevices; + + HRESULT hr; + bool validDevice = false; + if ( probeInfo.isInput == true ) { + DSCCAPS caps; + LPDIRECTSOUNDCAPTURE object; + + hr = DirectSoundCaptureCreate( lpguid, &object, NULL ); + if ( hr != DS_OK ) return TRUE; + + caps.dwSize = sizeof(caps); + hr = object->GetCaps( &caps ); + if ( hr == DS_OK ) { + if ( caps.dwChannels > 0 && caps.dwFormats > 0 ) + validDevice = true; + } + object->Release(); + } + else { + DSCAPS caps; + LPDIRECTSOUND object; + hr = DirectSoundCreate( lpguid, &object, NULL ); + if ( hr != DS_OK ) return TRUE; + + caps.dwSize = sizeof(caps); + hr = object->GetCaps( &caps ); + if ( hr == DS_OK ) { + if ( caps.dwFlags & DSCAPS_PRIMARYMONO || caps.dwFlags & DSCAPS_PRIMARYSTEREO ) + validDevice = true; + } + object->Release(); + } + + // If good device, then save its name and guid. + std::string name = convertCharPointerToStdString( description ); + //if ( name == "Primary Sound Driver" || name == "Primary Sound Capture Driver" ) + if ( lpguid == NULL ) + name = "Default Device"; + if ( validDevice ) { + for ( unsigned int i=0; i<dsDevices.size(); i++ ) { + if ( dsDevices[i].name == name ) { + dsDevices[i].found = true; + if ( probeInfo.isInput ) { + dsDevices[i].id[1] = lpguid; + dsDevices[i].validId[1] = true; + } + else { + dsDevices[i].id[0] = lpguid; + dsDevices[i].validId[0] = true; + } + return TRUE; + } + } + + DsDevice device; + device.name = name; + device.found = true; + if ( probeInfo.isInput ) { + device.id[1] = lpguid; + device.validId[1] = true; + } + else { + device.id[0] = lpguid; + device.validId[0] = true; + } + dsDevices.push_back( device ); + } + + return TRUE; +} + +static const char* getErrorString( int code ) +{ + switch ( code ) { + + case DSERR_ALLOCATED: + return "Already allocated"; + + case DSERR_CONTROLUNAVAIL: + return "Control unavailable"; + + case DSERR_INVALIDPARAM: + return "Invalid parameter"; + + case DSERR_INVALIDCALL: + return "Invalid call"; + + case DSERR_GENERIC: + return "Generic error"; + + case DSERR_PRIOLEVELNEEDED: + return "Priority level needed"; + + case DSERR_OUTOFMEMORY: + return "Out of memory"; + + case DSERR_BADFORMAT: + return "The sample rate or the channel format is not supported"; + + case DSERR_UNSUPPORTED: + return "Not supported"; + + case DSERR_NODRIVER: + return "No driver"; + + case DSERR_ALREADYINITIALIZED: + return "Already initialized"; + + case DSERR_NOAGGREGATION: + return "No aggregation"; + + case DSERR_BUFFERLOST: + return "Buffer lost"; + + case DSERR_OTHERAPPHASPRIO: + return "Another application already has priority"; + + case DSERR_UNINITIALIZED: + return "Uninitialized"; + + default: + return "DirectSound unknown error"; + } +} +//******************** End of __WINDOWS_DS__ *********************// +#endif + + +#if defined(__LINUX_ALSA__) + +#include <alsa/asoundlib.h> +#include <unistd.h> + + // A structure to hold various information related to the ALSA API + // implementation. +struct AlsaHandle { + snd_pcm_t *handles[2]; + bool synchronized; + bool xrun[2]; + pthread_cond_t runnable_cv; + bool runnable; + + AlsaHandle() + :synchronized(false), runnable(false) { xrun[0] = false; xrun[1] = false; } +}; + +static void *alsaCallbackHandler( void * ptr ); + +RtApiAlsa :: RtApiAlsa() +{ + // Nothing to do here. +} + +RtApiAlsa :: ~RtApiAlsa() +{ + if ( stream_.state != STREAM_CLOSED ) closeStream(); +} + +unsigned int RtApiAlsa :: getDeviceCount( void ) +{ + unsigned nDevices = 0; + int result, subdevice, card; + char name[64]; + snd_ctl_t *handle = 0; + + strcpy(name, "default"); + result = snd_ctl_open( &handle, "default", 0 ); + if (result == 0) { + nDevices++; + snd_ctl_close( 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 ) { + handle = 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: + if ( handle ) + snd_ctl_close( handle ); + snd_card_next( &card ); + } + + return nDevices; +} + +RtAudio::DeviceInfo RtApiAlsa :: getDeviceInfo( unsigned int device ) +{ + RtAudio::DeviceInfo info; + info.probed = false; + + unsigned nDevices = 0; + int result=-1, subdevice=-1, card=-1; + char name[64]; + snd_ctl_t *chandle = 0; + + result = snd_ctl_open( &chandle, "default", SND_CTL_NONBLOCK ); + if ( result == 0 ) { + if ( nDevices++ == device ) { + strcpy( name, "default" ); + goto foundDevice; + } + } + + // Count cards and devices + snd_card_next( &card ); + while ( card >= 0 ) { + sprintf( name, "hw:%d", card ); + result = snd_ctl_open( &chandle, name, SND_CTL_NONBLOCK ); + if ( result < 0 ) { + chandle = 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: + if ( chandle ) + snd_ctl_close( chandle ); + snd_card_next( &card ); + } + + if ( nDevices == 0 ) { + errorText_ = "RtApiAlsa::getDeviceInfo: no devices found!"; + error( RtAudioError::INVALID_USE ); + return info; + } + + if ( device >= nDevices ) { + errorText_ = "RtApiAlsa::getDeviceInfo: device ID is invalid!"; + error( RtAudioError::INVALID_USE ); + return info; + } + + foundDevice: + + // If a stream is already open, we cannot probe the stream devices. + // Thus, use the saved results. + if ( stream_.state != STREAM_CLOSED && + ( stream_.device[0] == device || stream_.device[1] == device ) ) { + snd_ctl_close( chandle ); + if ( device >= devices_.size() ) { + errorText_ = "RtApiAlsa::getDeviceInfo: device ID was not present before stream was opened."; + error( RtAudioError::WARNING ); + return info; + } + return devices_[ device ]; + } + + int openMode = SND_PCM_ASYNC; + snd_pcm_stream_t stream; + snd_pcm_info_t *pcminfo; + snd_pcm_info_alloca( &pcminfo ); + snd_pcm_t *phandle; + snd_pcm_hw_params_t *params; + snd_pcm_hw_params_alloca( ¶ms ); + + // First try for playback unless default device (which has subdev -1) + stream = SND_PCM_STREAM_PLAYBACK; + snd_pcm_info_set_stream( pcminfo, stream ); + if ( subdevice != -1 ) { + snd_pcm_info_set_device( pcminfo, subdevice ); + snd_pcm_info_set_subdevice( pcminfo, 0 ); + + result = snd_ctl_pcm_info( chandle, pcminfo ); + if ( result < 0 ) { + // Device probably doesn't support playback. + goto captureProbe; + } + } + + result = snd_pcm_open( &phandle, name, stream, openMode | SND_PCM_NONBLOCK ); + if ( result < 0 ) { + errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_open error for device (" << name << "), " << snd_strerror( result ) << "."; + errorText_ = errorStream_.str(); + error( RtAudioError::WARNING ); + goto captureProbe; + } + + // The device is open ... fill the parameter structure. + result = snd_pcm_hw_params_any( phandle, params ); + if ( result < 0 ) { + snd_pcm_close( phandle ); + errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_hw_params error for device (" << name << "), " << snd_strerror( result ) << "."; + errorText_ = errorStream_.str(); + error( RtAudioError::WARNING ); + goto captureProbe; + } + + // Get output channel information. + unsigned int value; + result = snd_pcm_hw_params_get_channels_max( params, &value ); + if ( result < 0 ) { + snd_pcm_close( phandle ); + errorStream_ << "RtApiAlsa::getDeviceInfo: error getting device (" << name << ") output channels, " << snd_strerror( result ) << "."; + errorText_ = errorStream_.str(); + error( RtAudioError::WARNING ); + goto captureProbe; + } + info.outputChannels = value; + snd_pcm_close( phandle ); + + captureProbe: + stream = SND_PCM_STREAM_CAPTURE; + snd_pcm_info_set_stream( pcminfo, stream ); + + // Now try for capture unless default device (with subdev = -1) + if ( subdevice != -1 ) { + result = snd_ctl_pcm_info( chandle, pcminfo ); + snd_ctl_close( chandle ); + if ( result < 0 ) { + // Device probably doesn't support capture. + if ( info.outputChannels == 0 ) return info; + goto probeParameters; + } + } + else + snd_ctl_close( chandle ); + + result = snd_pcm_open( &phandle, name, stream, openMode | SND_PCM_NONBLOCK); + if ( result < 0 ) { + errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_open error for device (" << name << "), " << snd_strerror( result ) << "."; + errorText_ = errorStream_.str(); + error( RtAudioError::WARNING ); + if ( info.outputChannels == 0 ) return info; + goto probeParameters; + } + + // The device is open ... fill the parameter structure. + result = snd_pcm_hw_params_any( phandle, params ); + if ( result < 0 ) { + snd_pcm_close( phandle ); + errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_hw_params error for device (" << name << "), " << snd_strerror( result ) << "."; + errorText_ = errorStream_.str(); + error( RtAudioError::WARNING ); + if ( info.outputChannels == 0 ) return info; + goto probeParameters; + } + + result = snd_pcm_hw_params_get_channels_max( params, &value ); + if ( result < 0 ) { + snd_pcm_close( phandle ); + errorStream_ << "RtApiAlsa::getDeviceInfo: error getting device (" << name << ") input channels, " << snd_strerror( result ) << "."; + errorText_ = errorStream_.str(); + error( RtAudioError::WARNING ); + if ( info.outputChannels == 0 ) return info; + goto probeParameters; + } + info.inputChannels = value; + snd_pcm_close( phandle ); + + // If device opens for both playback and capture, we determine the channels. + if ( info.outputChannels > 0 && info.inputChannels > 0 ) + info.duplexChannels = (info.outputChannels > info.inputChannels) ? info.inputChannels : info.outputChannels; + + // ALSA doesn't provide default devices so we'll use the first available one. + if ( device == 0 && info.outputChannels > 0 ) + info.isDefaultOutput = true; + if ( device == 0 && info.inputChannels > 0 ) + info.isDefaultInput = true; + + probeParameters: + // At this point, we just need to figure out the supported data + // formats and sample rates. We'll proceed by opening the device in + // the direction with the maximum number of channels, or playback if + // they are equal. This might limit our sample rate options, but so + // be it. + + if ( info.outputChannels >= info.inputChannels ) + stream = SND_PCM_STREAM_PLAYBACK; + else + stream = SND_PCM_STREAM_CAPTURE; + snd_pcm_info_set_stream( pcminfo, stream ); + + result = snd_pcm_open( &phandle, name, stream, openMode | SND_PCM_NONBLOCK); + if ( result < 0 ) { + errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_open error for device (" << name << "), " << snd_strerror( result ) << "."; + errorText_ = errorStream_.str(); + error( RtAudioError::WARNING ); + return info; + } + + // The device is open ... fill the parameter structure. + result = snd_pcm_hw_params_any( phandle, params ); + if ( result < 0 ) { + snd_pcm_close( phandle ); + errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_hw_params error for device (" << name << "), " << snd_strerror( result ) << "."; + errorText_ = errorStream_.str(); + error( RtAudioError::WARNING ); + return info; + } + + // Test our discrete set of sample rate values. + info.sampleRates.clear(); + for ( unsigned int i=0; i<MAX_SAMPLE_RATES; i++ ) { + if ( snd_pcm_hw_params_test_rate( phandle, params, SAMPLE_RATES[i], 0 ) == 0 ) { + info.sampleRates.push_back( SAMPLE_RATES[i] ); + + if ( !info.preferredSampleRate || ( SAMPLE_RATES[i] <= 48000 && SAMPLE_RATES[i] > info.preferredSampleRate ) ) + info.preferredSampleRate = SAMPLE_RATES[i]; + } + } + if ( info.sampleRates.size() == 0 ) { + snd_pcm_close( phandle ); + errorStream_ << "RtApiAlsa::getDeviceInfo: no supported sample rates found for device (" << name << ")."; + errorText_ = errorStream_.str(); + error( RtAudioError::WARNING ); + return info; + } + + // Probe the supported data formats ... we don't care about endian-ness just yet + snd_pcm_format_t format; + info.nativeFormats = 0; + format = SND_PCM_FORMAT_S8; + if ( snd_pcm_hw_params_test_format( phandle, params, format ) == 0 ) + info.nativeFormats |= RTAUDIO_SINT8; + format = SND_PCM_FORMAT_S16; + if ( snd_pcm_hw_params_test_format( phandle, params, format ) == 0 ) + info.nativeFormats |= RTAUDIO_SINT16; + format = SND_PCM_FORMAT_S24; + if ( snd_pcm_hw_params_test_format( phandle, params, format ) == 0 ) + info.nativeFormats |= RTAUDIO_SINT24; + format = SND_PCM_FORMAT_S32; + if ( snd_pcm_hw_params_test_format( phandle, params, format ) == 0 ) + info.nativeFormats |= RTAUDIO_SINT32; + format = SND_PCM_FORMAT_FLOAT; + if ( snd_pcm_hw_params_test_format( phandle, params, format ) == 0 ) + info.nativeFormats |= RTAUDIO_FLOAT32; + format = SND_PCM_FORMAT_FLOAT64; + if ( snd_pcm_hw_params_test_format( phandle, params, format ) == 0 ) + info.nativeFormats |= RTAUDIO_FLOAT64; + + // Check that we have at least one supported format + if ( info.nativeFormats == 0 ) { + snd_pcm_close( phandle ); + errorStream_ << "RtApiAlsa::getDeviceInfo: pcm device (" << name << ") data format not supported by RtAudio."; + errorText_ = errorStream_.str(); + error( RtAudioError::WARNING ); + return info; + } + + // Get the device name + if (strncmp(name, "default", 7)!=0) { + 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__) + 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 + + // 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 ( device == 0 + || (options && options->flags & RTAUDIO_ALSA_USE_DEFAULT) ) + { + strcpy(name, "default"); + result = snd_ctl_open( &chandle, "default", SND_CTL_NONBLOCK ); + if ( result == 0 ) { + if ( nDevices == device ) { + strcpy( name, "default" ); + snd_ctl_close( chandle ); + goto foundDevice; + } + nDevices++; + } + } + + else { + nDevices++; + // 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 ); + } + + 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 (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, ¶m); + } + 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( 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 ); + + #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( RtAudioError::SYSTEM_ERROR ); +} + +void RtApiAlsa :: stopStream() +{ + verifyStream(); + if ( stream_.state == STREAM_STOPPED ) { + errorText_ = "RtApiAlsa::stopStream(): the stream is already stopped!"; + error( RtAudioError::WARNING ); + return; + } + + stream_.state = STREAM_STOPPED; + MUTEX_LOCK( &stream_.mutex ); + + int result = 0; + AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle; + snd_pcm_t **handle = (snd_pcm_t **) apiInfo->handles; + if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { + if ( apiInfo->synchronized ) + result = snd_pcm_drop( handle[0] ); + else + result = snd_pcm_drain( handle[0] ); + if ( result < 0 ) { + errorStream_ << "RtApiAlsa::stopStream: error draining output pcm device, " << snd_strerror( result ) << "."; + errorText_ = errorStream_.str(); + goto unlock; + } + } + + if ( ( stream_.mode == INPUT || stream_.mode == DUPLEX ) && !apiInfo->synchronized ) { + result = snd_pcm_drop( handle[1] ); + if ( result < 0 ) { + errorStream_ << "RtApiAlsa::stopStream: error stopping input pcm device, " << snd_strerror( result ) << "."; + errorText_ = errorStream_.str(); + goto unlock; + } + } + + unlock: + apiInfo->runnable = false; // fixes high CPU usage when stopped + MUTEX_UNLOCK( &stream_.mutex ); + + if ( result >= 0 ) return; + error( RtAudioError::SYSTEM_ERROR ); +} + +void RtApiAlsa :: abortStream() +{ + verifyStream(); + if ( stream_.state == STREAM_STOPPED ) { + errorText_ = "RtApiAlsa::abortStream(): the stream is already stopped!"; + error( RtAudioError::WARNING ); + return; + } + + stream_.state = STREAM_STOPPED; + MUTEX_LOCK( &stream_.mutex ); + + int result = 0; + AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle; + snd_pcm_t **handle = (snd_pcm_t **) apiInfo->handles; + if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { + result = snd_pcm_drop( handle[0] ); + if ( result < 0 ) { + errorStream_ << "RtApiAlsa::abortStream: error aborting output pcm device, " << snd_strerror( result ) << "."; + errorText_ = errorStream_.str(); + goto unlock; + } + } + + if ( ( stream_.mode == INPUT || stream_.mode == DUPLEX ) && !apiInfo->synchronized ) { + result = snd_pcm_drop( handle[1] ); + if ( result < 0 ) { + errorStream_ << "RtApiAlsa::abortStream: error aborting input pcm device, " << snd_strerror( result ) << "."; + errorText_ = errorStream_.str(); + goto unlock; + } + } + + unlock: + apiInfo->runnable = false; // fixes high CPU usage when stopped + MUTEX_UNLOCK( &stream_.mutex ); + + if ( result >= 0 ) return; + error( RtAudioError::SYSTEM_ERROR ); +} + +void RtApiAlsa :: callbackEvent() +{ + AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle; + if ( stream_.state == STREAM_STOPPED ) { + MUTEX_LOCK( &stream_.mutex ); + while ( !apiInfo->runnable ) + pthread_cond_wait( &apiInfo->runnable_cv, &stream_.mutex ); + + if ( stream_.state != STREAM_RUNNING ) { + MUTEX_UNLOCK( &stream_.mutex ); + return; + } + MUTEX_UNLOCK( &stream_.mutex ); + } + + if ( stream_.state == STREAM_CLOSED ) { + errorText_ = "RtApiAlsa::callbackEvent(): the stream is closed ... this shouldn't happen!"; + error( RtAudioError::WARNING ); + return; + } + + int doStopStream = 0; + RtAudioCallback callback = (RtAudioCallback) stream_.callbackInfo.callback; + double streamTime = getStreamTime(); + RtAudioStreamStatus status = 0; + if ( stream_.mode != INPUT && apiInfo->xrun[0] == true ) { + status |= RTAUDIO_OUTPUT_UNDERFLOW; + apiInfo->xrun[0] = false; + } + if ( stream_.mode != OUTPUT && apiInfo->xrun[1] == true ) { + status |= RTAUDIO_INPUT_OVERFLOW; + apiInfo->xrun[1] = false; + } + doStopStream = callback( stream_.userBuffer[0], stream_.userBuffer[1], + stream_.bufferSize, streamTime, status, stream_.callbackInfo.userData ); + + if ( doStopStream == 2 ) { + abortStream(); + return; + } + + MUTEX_LOCK( &stream_.mutex ); + + // The state might change while waiting on a mutex. + if ( stream_.state == STREAM_STOPPED ) goto unlock; + + int result; + char *buffer; + int channels; + snd_pcm_t **handle; + snd_pcm_sframes_t frames; + RtAudioFormat format; + handle = (snd_pcm_t **) apiInfo->handles; + + if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) { + + // Setup parameters. + if ( stream_.doConvertBuffer[1] ) { + buffer = stream_.deviceBuffer; + channels = stream_.nDeviceChannels[1]; + format = stream_.deviceFormat[1]; + } + else { + buffer = stream_.userBuffer[1]; + channels = stream_.nUserChannels[1]; + format = stream_.userFormat; + } + + // Read samples from device in interleaved/non-interleaved format. + if ( stream_.deviceInterleaved[1] ) + result = snd_pcm_readi( handle[1], buffer, stream_.bufferSize ); + else { + void *bufs[channels]; + size_t offset = stream_.bufferSize * formatBytes( format ); + for ( int i=0; i<channels; i++ ) + bufs[i] = (void *) (buffer + (i * offset)); + result = snd_pcm_readn( handle[1], bufs, stream_.bufferSize ); + } + + if ( result < (int) stream_.bufferSize ) { + // Either an error or overrun occured. + if ( result == -EPIPE ) { + snd_pcm_state_t state = snd_pcm_state( handle[1] ); + if ( state == SND_PCM_STATE_XRUN ) { + apiInfo->xrun[1] = true; + result = snd_pcm_prepare( handle[1] ); + if ( result < 0 ) { + errorStream_ << "RtApiAlsa::callbackEvent: error preparing device after overrun, " << snd_strerror( result ) << "."; + errorText_ = errorStream_.str(); + } + } + else { + errorStream_ << "RtApiAlsa::callbackEvent: error, current state is " << snd_pcm_state_name( state ) << ", " << snd_strerror( result ) << "."; + errorText_ = errorStream_.str(); + } + } + else { + errorStream_ << "RtApiAlsa::callbackEvent: audio read error, " << snd_strerror( result ) << "."; + errorText_ = errorStream_.str(); + } + error( RtAudioError::WARNING ); + goto tryOutput; + } + + // Do byte swapping if necessary. + if ( stream_.doByteSwap[1] ) + byteSwapBuffer( buffer, stream_.bufferSize * channels, format ); + + // Do buffer conversion if necessary. + if ( stream_.doConvertBuffer[1] ) + convertBuffer( stream_.userBuffer[1], stream_.deviceBuffer, stream_.convertInfo[1] ); + + // Check stream latency + result = snd_pcm_delay( handle[1], &frames ); + if ( result == 0 && frames > 0 ) stream_.latency[1] = frames; + } + + tryOutput: + + if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { + + // Setup parameters and do buffer conversion if necessary. + if ( stream_.doConvertBuffer[0] ) { + buffer = stream_.deviceBuffer; + convertBuffer( buffer, stream_.userBuffer[0], stream_.convertInfo[0] ); + channels = stream_.nDeviceChannels[0]; + format = stream_.deviceFormat[0]; + } + else { + buffer = stream_.userBuffer[0]; + channels = stream_.nUserChannels[0]; + format = stream_.userFormat; + } + + // Do byte swapping if necessary. + if ( stream_.doByteSwap[0] ) + byteSwapBuffer(buffer, stream_.bufferSize * channels, format); + + // Write samples to device in interleaved/non-interleaved format. + if ( stream_.deviceInterleaved[0] ) + result = snd_pcm_writei( handle[0], buffer, stream_.bufferSize ); + else { + void *bufs[channels]; + size_t offset = stream_.bufferSize * formatBytes( format ); + for ( int i=0; i<channels; i++ ) + bufs[i] = (void *) (buffer + (i * offset)); + result = snd_pcm_writen( handle[0], bufs, stream_.bufferSize ); + } + + if ( result < (int) stream_.bufferSize ) { + // Either an error or underrun occured. + if ( result == -EPIPE ) { + snd_pcm_state_t state = snd_pcm_state( handle[0] ); + if ( state == SND_PCM_STATE_XRUN ) { + apiInfo->xrun[0] = true; + result = snd_pcm_prepare( handle[0] ); + if ( result < 0 ) { + errorStream_ << "RtApiAlsa::callbackEvent: error preparing device after underrun, " << snd_strerror( result ) << "."; + errorText_ = errorStream_.str(); + } + else + errorText_ = "RtApiAlsa::callbackEvent: audio write error, underrun."; + } + else { + errorStream_ << "RtApiAlsa::callbackEvent: error, current state is " << snd_pcm_state_name( state ) << ", " << snd_strerror( result ) << "."; + errorText_ = errorStream_.str(); + } + } + else { + errorStream_ << "RtApiAlsa::callbackEvent: audio write error, " << snd_strerror( result ) << "."; + errorText_ = errorStream_.str(); + } + error( RtAudioError::WARNING ); + goto unlock; + } + + // Check stream latency + result = snd_pcm_delay( handle[0], &frames ); + if ( result == 0 && frames > 0 ) stream_.latency[0] = frames; + } + + unlock: + MUTEX_UNLOCK( &stream_.mutex ); + + RtApi::tickStreamTime(); + if ( doStopStream == 1 ) this->stopStream(); +} + +static void *alsaCallbackHandler( void *ptr ) +{ + CallbackInfo *info = (CallbackInfo *) ptr; + RtApiAlsa *object = (RtApiAlsa *) info->object; + bool *isRunning = &info->isRunning; + +#ifdef SCHED_RR // Undefined with some OSes (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. + +#include <pulse/error.h> +#include <pulse/simple.h> +#include <pulse/pulseaudio.h> +#include <cstdio> + +static pa_mainloop_api *rt_pa_mainloop_api = NULL; +struct PaDeviceInfo { + PaDeviceInfo() : sink_index(-1), source_index(-1) {} + int sink_index; + int source_index; + std::string sink_name; + std::string source_name; + RtAudio::DeviceInfo info; +}; +static struct { + std::vector<PaDeviceInfo> dev; + std::string default_sink_name; + std::string default_source_name; + int default_rate; +} rt_pa_info; + +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_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) { } +}; + +static void rt_pa_mainloop_api_quit(int ret) { + rt_pa_mainloop_api->quit(rt_pa_mainloop_api, ret); +} + +static void rt_pa_server_callback(pa_context *context, const pa_server_info *info, void *data){ + (void)context; + (void)data; + pa_sample_spec ss; + + if (!info) + rt_pa_mainloop_api_quit(1); + + ss = info->sample_spec; + + rt_pa_info.default_rate = ss.rate; + rt_pa_info.default_sink_name = info->default_sink_name; + rt_pa_info.default_source_name = info->default_source_name; + rt_pa_mainloop_api_quit(0); +} + +static void rt_pa_sink_info_cb(pa_context * /*c*/, const pa_sink_info *i, + int eol, void * /*userdata*/) +{ + if (eol) return; + PaDeviceInfo inf; + inf.info.name = pa_proplist_gets(i->proplist, "device.description"); + inf.info.probed = true; + inf.info.outputChannels = i->sample_spec.channels; + inf.info.preferredSampleRate = i->sample_spec.rate; + inf.info.isDefaultOutput = (rt_pa_info.default_sink_name == i->name); + inf.sink_index = i->index; + inf.sink_name = i->name; + for ( const unsigned int *sr = SUPPORTED_SAMPLERATES; *sr; ++sr ) + inf.info.sampleRates.push_back( *sr ); + for ( const rtaudio_pa_format_mapping_t *fm = supported_sampleformats; + fm->rtaudio_format; ++fm ) + inf.info.nativeFormats |= fm->rtaudio_format; + for (size_t i=0; i < rt_pa_info.dev.size(); i++) + { + /* Attempt to match up sink and source records by device description. */ + if (rt_pa_info.dev[i].info.name == inf.info.name) { + rt_pa_info.dev[i].sink_index = inf.sink_index; + rt_pa_info.dev[i].sink_name = inf.sink_name; + rt_pa_info.dev[i].info.outputChannels = inf.info.outputChannels; + rt_pa_info.dev[i].info.isDefaultOutput = inf.info.isDefaultOutput; + /* Assume duplex channels are minimum of input and output channels. */ + /* Uncomment if we add support for DUPLEX + if (rt_pa_info.dev[i].source_index > -1) + (inf.info.outputChannels < rt_pa_info.dev[i].info.inputChannels) + ? inf.info.outputChannels : rt_pa_info.dev[i].info.inputChannels; + */ + return; + } + } + /* try to ensure device #0 is the default */ + if (inf.info.isDefaultOutput) + rt_pa_info.dev.insert(rt_pa_info.dev.begin(), inf); + else + rt_pa_info.dev.push_back(inf); +} + +static void rt_pa_source_info_cb(pa_context * /*c*/, const pa_source_info *i, + int eol, void * /*userdata*/) +{ + if (eol) return; + PaDeviceInfo inf; + inf.info.name = pa_proplist_gets(i->proplist, "device.description"); + inf.info.probed = true; + inf.info.inputChannels = i->sample_spec.channels; + inf.info.preferredSampleRate = i->sample_spec.rate; + inf.info.isDefaultInput = (rt_pa_info.default_source_name == i->name); + inf.source_index = i->index; + inf.source_name = i->name; + for ( const unsigned int *sr = SUPPORTED_SAMPLERATES; *sr; ++sr ) + inf.info.sampleRates.push_back( *sr ); + for ( const rtaudio_pa_format_mapping_t *fm = supported_sampleformats; + fm->rtaudio_format; ++fm ) + inf.info.nativeFormats |= fm->rtaudio_format; + + for (size_t i=0; i < rt_pa_info.dev.size(); i++) + { + /* Attempt to match up sink and source records by device description. */ + if (rt_pa_info.dev[i].info.name == inf.info.name) { + rt_pa_info.dev[i].source_index = inf.source_index; + rt_pa_info.dev[i].source_name = inf.source_name; + rt_pa_info.dev[i].info.inputChannels = inf.info.inputChannels; + rt_pa_info.dev[i].info.isDefaultInput = inf.info.isDefaultInput; + /* Assume duplex channels are minimum of input and output channels. */ + /* Uncomment if we add support for DUPLEX + if (rt_pa_info.dev[i].sink_index > -1) { + rt_pa_info.dev[i].info.duplexChannels = + (inf.info.inputChannels < rt_pa_info.dev[i].info.outputChannels) + ? inf.info.inputChannels : rt_pa_info.dev[i].info.outputChannels; + } + */ + return; + } + } + /* try to ensure device #0 is the default */ + if (inf.info.isDefaultInput) + rt_pa_info.dev.insert(rt_pa_info.dev.begin(), inf); + else + rt_pa_info.dev.push_back(inf); +} + +static void rt_pa_context_state_callback(pa_context *context, void *userdata) { + (void)userdata; + + switch (pa_context_get_state(context)) { + case PA_CONTEXT_CONNECTING: + case PA_CONTEXT_AUTHORIZING: + case PA_CONTEXT_SETTING_NAME: + break; + + case PA_CONTEXT_READY: + rt_pa_info.dev.clear(); + pa_context_get_server_info(context, rt_pa_server_callback, NULL); + pa_context_get_sink_info_list(context, rt_pa_sink_info_cb, NULL); + pa_context_get_source_info_list(context, rt_pa_source_info_cb, NULL); + break; + + case PA_CONTEXT_TERMINATED: + rt_pa_mainloop_api_quit(0); + break; + + case PA_CONTEXT_FAILED: + default: + rt_pa_mainloop_api_quit(1); + } +} + +RtApiPulse::~RtApiPulse() +{ + if ( stream_.state != STREAM_CLOSED ) + closeStream(); +} + +void RtApiPulse::collectDeviceInfo( void ) +{ + pa_context *context = NULL; + pa_mainloop *m = NULL; + char *server = NULL; + int ret = 1; + + if (!(m = pa_mainloop_new())) { + errorStream_ << "RtApiPulse::DeviceInfo pa_mainloop_new() failed."; + errorText_ = errorStream_.str(); + error( RtAudioError::WARNING ); + goto quit; + } + + rt_pa_mainloop_api = pa_mainloop_get_api(m); + + if (!(context = pa_context_new_with_proplist(rt_pa_mainloop_api, NULL, NULL))) { + errorStream_ << "pa_context_new() failed."; + errorText_ = errorStream_.str(); + error( RtAudioError::WARNING ); + goto quit; + } + + pa_context_set_state_callback(context, rt_pa_context_state_callback, NULL); + + if (pa_context_connect(context, server, PA_CONTEXT_NOFLAGS, NULL) < 0) { + errorStream_ << "RtApiPulse::DeviceInfo pa_context_connect() failed: " + << pa_strerror(pa_context_errno(context)); + errorText_ = errorStream_.str(); + error( RtAudioError::WARNING ); + goto quit; + } + + if (pa_mainloop_run(m, &ret) < 0) { + errorStream_ << "pa_mainloop_run() failed."; + errorText_ = errorStream_.str(); + error( RtAudioError::WARNING ); + goto quit; + } + +quit: + if (context) + pa_context_unref(context); + + if (m) { + pa_mainloop_free(m); + } + + pa_xfree(server); +} + +unsigned int RtApiPulse::getDeviceCount( void ) +{ + collectDeviceInfo(); + return rt_pa_info.dev.size(); +} + +RtAudio::DeviceInfo RtApiPulse::getDeviceInfo( unsigned int device ) +{ + if (rt_pa_info.dev.size()==0) + collectDeviceInfo(); + if (device < rt_pa_info.dev.size()) + return rt_pa_info.dev[device].info; + return RtAudio::DeviceInfo(); +} + +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 ); +} + +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 ); + + #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 ); +} + +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->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 ); + 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->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 ); + 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 >= rt_pa_info.dev.size() ) return false; + if ( firstChannel != 0 ) { + errorText_ = "PulseAudio does not support channel offset mapping."; + return false; + } + + /* these may be NULL for default, but we've already got the names */ + const char *dev_input = NULL; + const char *dev_output = NULL; + if (!rt_pa_info.dev[device].source_name.empty()) + dev_input = rt_pa_info.dev[device].source_name.c_str(); + if (!rt_pa_info.dev[device].sink_name.empty()) + dev_output = rt_pa_info.dev[device].sink_name.c_str(); + + if (mode==INPUT && rt_pa_info.dev[device].info.inputChannels == 0) { + errorText_ = "PulseAudio device does not support input."; + return false; + } + if (mode==OUTPUT && rt_pa_info.dev[device].info.outputChannels == 0) { + errorText_ = "PulseAudio device does not support output."; + return false; + } + if (mode==DUPLEX && rt_pa_info.dev[device].info.duplexChannels == 0) { + /* Note: will always error, DUPLEX not yet supported */ + errorText_ = "PulseAudio device does not support duplex."; + return false; + } + + if (mode==INPUT && rt_pa_info.dev[device].info.inputChannels < channels) { + errorText_ = "PulseAudio: unsupported number of input channels."; + return false; + } + + if (mode==OUTPUT && rt_pa_info.dev[device].info.outputChannels < channels) { + errorText_ = "PulseAudio: unsupported number of output channels."; + return false; + } + + if (mode==DUPLEX && rt_pa_info.dev[device].info.duplexChannels < channels) { + /* Note: will always error, DUPLEX not yet supported */ + errorText_ = "PulseAudio: unsupported number of duplex channels."; + 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 ) { + 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; + 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_.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, + 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: + pah->s_play = pa_simple_new( NULL, streamName.c_str(), PA_STREAM_PLAYBACK, + dev_output, "Playback", &ss, NULL, NULL, &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, ¶m); + } + 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; +} + +//******************** End of __LINUX_PULSE__ *********************// +#endif + +#if defined(__LINUX_OSS__) + +#include <unistd.h> +#include <sys/ioctl.h> +#include <unistd.h> +#include <fcntl.h> +#include <sys/soundcard.h> +#include <errno.h> +#include <math.h> + +static void *ossCallbackHandler(void * ptr); + +// A structure to hold various information related to the OSS API +// implementation. +struct OssHandle { + int id[2]; // device ids + bool xrun[2]; + bool triggered; + pthread_cond_t runnable; + + OssHandle() + :triggered(false) { id[0] = 0; id[1] = 0; xrun[0] = false; xrun[1] = false; } +}; + +RtApiOss :: RtApiOss() +{ + // Nothing to do here. +} + +RtApiOss :: ~RtApiOss() +{ + if ( stream_.state != STREAM_CLOSED ) closeStream(); +} + +unsigned int RtApiOss :: getDeviceCount( void ) +{ + int mixerfd = open( "/dev/mixer", O_RDWR, 0 ); + if ( mixerfd == -1 ) { + errorText_ = "RtApiOss::getDeviceCount: error opening '/dev/mixer'."; + error( RtAudioError::WARNING ); + return 0; + } + + oss_sysinfo sysinfo; + if ( ioctl( mixerfd, SNDCTL_SYSINFO, &sysinfo ) == -1 ) { + close( mixerfd ); + errorText_ = "RtApiOss::getDeviceCount: error getting sysinfo, OSS version >= 4.0 is required."; + error( RtAudioError::WARNING ); + return 0; + } + + close( mixerfd ); + return sysinfo.numaudios; +} + +RtAudio::DeviceInfo RtApiOss :: getDeviceInfo( unsigned int device ) +{ + RtAudio::DeviceInfo info; + info.probed = false; + + int mixerfd = open( "/dev/mixer", O_RDWR, 0 ); + if ( mixerfd == -1 ) { + errorText_ = "RtApiOss::getDeviceInfo: error opening '/dev/mixer'."; + error( RtAudioError::WARNING ); + return info; + } + + oss_sysinfo sysinfo; + int result = ioctl( mixerfd, SNDCTL_SYSINFO, &sysinfo ); + if ( result == -1 ) { + close( mixerfd ); + errorText_ = "RtApiOss::getDeviceInfo: error getting sysinfo, OSS version >= 4.0 is required."; + error( RtAudioError::WARNING ); + return info; + } + + unsigned nDevices = sysinfo.numaudios; + if ( nDevices == 0 ) { + close( mixerfd ); + errorText_ = "RtApiOss::getDeviceInfo: no devices found!"; + error( RtAudioError::INVALID_USE ); + return info; + } + + if ( device >= nDevices ) { + close( mixerfd ); + errorText_ = "RtApiOss::getDeviceInfo: device ID is invalid!"; + error( RtAudioError::INVALID_USE ); + return info; + } + + oss_audioinfo ainfo; + ainfo.dev = device; + result = ioctl( mixerfd, SNDCTL_AUDIOINFO, &ainfo ); + close( mixerfd ); + if ( result == -1 ) { + errorStream_ << "RtApiOss::getDeviceInfo: error getting device (" << ainfo.name << ") info."; + errorText_ = errorStream_.str(); + error( RtAudioError::WARNING ); + return info; + } + + // Probe channels + if ( ainfo.caps & PCM_CAP_OUTPUT ) info.outputChannels = ainfo.max_channels; + if ( ainfo.caps & PCM_CAP_INPUT ) info.inputChannels = ainfo.max_channels; + if ( ainfo.caps & PCM_CAP_DUPLEX ) { + if ( info.outputChannels > 0 && info.inputChannels > 0 && ainfo.caps & PCM_CAP_DUPLEX ) + info.duplexChannels = (info.outputChannels > info.inputChannels) ? info.inputChannels : info.outputChannels; + } + + // Probe data formats ... do for input + unsigned long mask = ainfo.iformats; + if ( mask & AFMT_S16_LE || mask & AFMT_S16_BE ) + info.nativeFormats |= RTAUDIO_SINT16; + if ( mask & AFMT_S8 ) + info.nativeFormats |= RTAUDIO_SINT8; + if ( mask & AFMT_S32_LE || mask & AFMT_S32_BE ) + info.nativeFormats |= RTAUDIO_SINT32; +#ifdef AFMT_FLOAT + if ( mask & AFMT_FLOAT ) + info.nativeFormats |= RTAUDIO_FLOAT32; +#endif + if ( mask & AFMT_S24_LE || mask & AFMT_S24_BE ) + info.nativeFormats |= RTAUDIO_SINT24; + + // Check that we have at least one supported format + if ( info.nativeFormats == 0 ) { + errorStream_ << "RtApiOss::getDeviceInfo: device (" << ainfo.name << ") data format not supported by RtAudio."; + errorText_ = errorStream_.str(); + error( RtAudioError::WARNING ); + return info; + } + + // Probe the supported sample rates. + info.sampleRates.clear(); + if ( ainfo.nrates ) { + for ( unsigned int i=0; i<ainfo.nrates; i++ ) { + for ( unsigned int k=0; k<MAX_SAMPLE_RATES; k++ ) { + if ( ainfo.rates[i] == SAMPLE_RATES[k] ) { + info.sampleRates.push_back( SAMPLE_RATES[k] ); + + if ( !info.preferredSampleRate || ( SAMPLE_RATES[k] <= 48000 && SAMPLE_RATES[k] > info.preferredSampleRate ) ) + info.preferredSampleRate = SAMPLE_RATES[k]; + + break; + } + } + } + } + else { + // Check min and max rate values; + for ( unsigned int k=0; k<MAX_SAMPLE_RATES; k++ ) { + if ( ainfo.min_rate <= (int) SAMPLE_RATES[k] && ainfo.max_rate >= (int) SAMPLE_RATES[k] ) { + info.sampleRates.push_back( SAMPLE_RATES[k] ); + + if ( !info.preferredSampleRate || ( SAMPLE_RATES[k] <= 48000 && SAMPLE_RATES[k] > info.preferredSampleRate ) ) + info.preferredSampleRate = SAMPLE_RATES[k]; + } + } + } + + if ( info.sampleRates.size() == 0 ) { + errorStream_ << "RtApiOss::getDeviceInfo: no supported sample rates found for device (" << ainfo.name << ")."; + errorText_ = errorStream_.str(); + error( RtAudioError::WARNING ); + } + else { + info.probed = true; + info.name = ainfo.name; + } + + return info; +} + + +bool RtApiOss :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels, + unsigned int firstChannel, unsigned int sampleRate, + RtAudioFormat format, unsigned int *bufferSize, + RtAudio::StreamOptions *options ) +{ + int mixerfd = open( "/dev/mixer", O_RDWR, 0 ); + if ( mixerfd == -1 ) { + errorText_ = "RtApiOss::probeDeviceOpen: error opening '/dev/mixer'."; + return FAILURE; + } + + oss_sysinfo sysinfo; + int result = ioctl( mixerfd, SNDCTL_SYSINFO, &sysinfo ); + if ( result == -1 ) { + close( mixerfd ); + errorText_ = "RtApiOss::probeDeviceOpen: error getting sysinfo, OSS version >= 4.0 is required."; + return FAILURE; + } + + unsigned nDevices = sysinfo.numaudios; + if ( nDevices == 0 ) { + // This should not happen because a check is made before this function is called. + close( mixerfd ); + errorText_ = "RtApiOss::probeDeviceOpen: no devices found!"; + return FAILURE; + } + + if ( device >= nDevices ) { + // This should not happen because a check is made before this function is called. + close( mixerfd ); + errorText_ = "RtApiOss::probeDeviceOpen: device ID is invalid!"; + return FAILURE; + } + + oss_audioinfo ainfo; + ainfo.dev = device; + result = ioctl( mixerfd, SNDCTL_AUDIOINFO, &ainfo ); + close( mixerfd ); + if ( result == -1 ) { + errorStream_ << "RtApiOss::getDeviceInfo: error getting device (" << ainfo.name << ") info."; + errorText_ = errorStream_.str(); + return FAILURE; + } + + // Check if device supports input or output + if ( ( mode == OUTPUT && !( ainfo.caps & PCM_CAP_OUTPUT ) ) || + ( mode == INPUT && !( ainfo.caps & PCM_CAP_INPUT ) ) ) { + if ( mode == OUTPUT ) + errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") does not support output."; + else + errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") does not support input."; + errorText_ = errorStream_.str(); + return FAILURE; + } + + int flags = 0; + OssHandle *handle = (OssHandle *) stream_.apiHandle; + if ( mode == OUTPUT ) + flags |= O_WRONLY; + else { // mode == INPUT + if (stream_.mode == OUTPUT && stream_.device[0] == device) { + // We just set the same device for playback ... close and reopen for duplex (OSS only). + close( handle->id[0] ); + handle->id[0] = 0; + if ( !( ainfo.caps & PCM_CAP_DUPLEX ) ) { + errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") does not support duplex mode."; + errorText_ = errorStream_.str(); + return FAILURE; + } + // Check that the number previously set channels is the same. + if ( stream_.nUserChannels[0] != channels ) { + errorStream_ << "RtApiOss::probeDeviceOpen: input/output channels must be equal for OSS duplex device (" << ainfo.name << ")."; + errorText_ = errorStream_.str(); + return FAILURE; + } + flags |= O_RDWR; + } + else + flags |= O_RDONLY; + } + + // Set exclusive access if specified. + if ( options && options->flags & RTAUDIO_HOG_DEVICE ) flags |= O_EXCL; + + // Try to open the device. + int fd; + fd = open( ainfo.devnode, flags, 0 ); + if ( fd == -1 ) { + if ( errno == EBUSY ) + errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") is busy."; + else + errorStream_ << "RtApiOss::probeDeviceOpen: error opening device (" << ainfo.name << ")."; + errorText_ = errorStream_.str(); + return FAILURE; + } + + // For duplex operation, specifically set this mode (this doesn't seem to work). + /* + if ( flags | O_RDWR ) { + result = ioctl( fd, SNDCTL_DSP_SETDUPLEX, NULL ); + if ( result == -1) { + errorStream_ << "RtApiOss::probeDeviceOpen: error setting duplex mode for device (" << ainfo.name << ")."; + errorText_ = errorStream_.str(); + return FAILURE; + } + } + */ + + // Check the device channel support. + stream_.nUserChannels[mode] = channels; + if ( ainfo.max_channels < (int)(channels + firstChannel) ) { + close( fd ); + errorStream_ << "RtApiOss::probeDeviceOpen: the device (" << ainfo.name << ") does not support requested channel parameters."; + errorText_ = errorStream_.str(); + return FAILURE; + } + + // Set the number of channels. + int deviceChannels = channels + firstChannel; + result = ioctl( fd, SNDCTL_DSP_CHANNELS, &deviceChannels ); + if ( result == -1 || deviceChannels < (int)(channels + firstChannel) ) { + close( fd ); + errorStream_ << "RtApiOss::probeDeviceOpen: error setting channel parameters on device (" << ainfo.name << ")."; + errorText_ = errorStream_.str(); + return FAILURE; + } + stream_.nDeviceChannels[mode] = deviceChannels; + + // Get the data format mask + int mask; + result = ioctl( fd, SNDCTL_DSP_GETFMTS, &mask ); + if ( result == -1 ) { + close( fd ); + errorStream_ << "RtApiOss::probeDeviceOpen: error getting device (" << ainfo.name << ") data formats."; + errorText_ = errorStream_.str(); + return FAILURE; + } + + // Determine how to set the device format. + stream_.userFormat = format; + int deviceFormat = -1; + stream_.doByteSwap[mode] = false; + if ( format == RTAUDIO_SINT8 ) { + if ( mask & AFMT_S8 ) { + deviceFormat = AFMT_S8; + stream_.deviceFormat[mode] = RTAUDIO_SINT8; + } + } + else if ( format == RTAUDIO_SINT16 ) { + if ( mask & AFMT_S16_NE ) { + deviceFormat = AFMT_S16_NE; + stream_.deviceFormat[mode] = RTAUDIO_SINT16; + } + else if ( mask & AFMT_S16_OE ) { + deviceFormat = AFMT_S16_OE; + stream_.deviceFormat[mode] = RTAUDIO_SINT16; + stream_.doByteSwap[mode] = true; + } + } + else if ( format == RTAUDIO_SINT24 ) { + if ( mask & AFMT_S24_NE ) { + deviceFormat = AFMT_S24_NE; + stream_.deviceFormat[mode] = RTAUDIO_SINT24; + } + else if ( mask & AFMT_S24_OE ) { + deviceFormat = AFMT_S24_OE; + stream_.deviceFormat[mode] = RTAUDIO_SINT24; + stream_.doByteSwap[mode] = true; + } + } + else if ( format == RTAUDIO_SINT32 ) { + if ( mask & AFMT_S32_NE ) { + deviceFormat = AFMT_S32_NE; + stream_.deviceFormat[mode] = RTAUDIO_SINT32; + } + else if ( mask & AFMT_S32_OE ) { + deviceFormat = AFMT_S32_OE; + stream_.deviceFormat[mode] = RTAUDIO_SINT32; + stream_.doByteSwap[mode] = true; + } + } + + if ( deviceFormat == -1 ) { + // The user requested format is not natively supported by the device. + if ( mask & AFMT_S16_NE ) { + deviceFormat = AFMT_S16_NE; + stream_.deviceFormat[mode] = RTAUDIO_SINT16; + } + else if ( mask & AFMT_S32_NE ) { + deviceFormat = AFMT_S32_NE; + stream_.deviceFormat[mode] = RTAUDIO_SINT32; + } + else if ( mask & AFMT_S24_NE ) { + deviceFormat = AFMT_S24_NE; + stream_.deviceFormat[mode] = RTAUDIO_SINT24; + } + else if ( mask & AFMT_S16_OE ) { + deviceFormat = AFMT_S16_OE; + stream_.deviceFormat[mode] = RTAUDIO_SINT16; + stream_.doByteSwap[mode] = true; + } + else if ( mask & AFMT_S32_OE ) { + deviceFormat = AFMT_S32_OE; + stream_.deviceFormat[mode] = RTAUDIO_SINT32; + stream_.doByteSwap[mode] = true; + } + else if ( mask & AFMT_S24_OE ) { + deviceFormat = AFMT_S24_OE; + stream_.deviceFormat[mode] = RTAUDIO_SINT24; + stream_.doByteSwap[mode] = true; + } + else if ( mask & AFMT_S8) { + deviceFormat = AFMT_S8; + stream_.deviceFormat[mode] = RTAUDIO_SINT8; + } + } + + if ( stream_.deviceFormat[mode] == 0 ) { + // This really shouldn't happen ... + close( fd ); + errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") data format not supported by RtAudio."; + errorText_ = errorStream_.str(); + return FAILURE; + } + + // Set the data format. + int temp = deviceFormat; + result = ioctl( fd, SNDCTL_DSP_SETFMT, &deviceFormat ); + if ( result == -1 || deviceFormat != temp ) { + close( fd ); + errorStream_ << "RtApiOss::probeDeviceOpen: error setting data format on device (" << ainfo.name << ")."; + errorText_ = errorStream_.str(); + return FAILURE; + } + + // Attempt to set the buffer size. According to OSS, the minimum + // number of buffers is two. The supposed minimum buffer size is 16 + // bytes, so that will be our lower bound. The argument to this + // call is in the form 0xMMMMSSSS (hex), where the buffer size (in + // bytes) is given as 2^SSSS and the number of buffers as 2^MMMM. + // We'll check the actual value used near the end of the setup + // procedure. + int ossBufferBytes = *bufferSize * formatBytes( stream_.deviceFormat[mode] ) * deviceChannels; + if ( ossBufferBytes < 16 ) ossBufferBytes = 16; + int buffers = 0; + if ( options ) buffers = options->numberOfBuffers; + if ( options && options->flags & RTAUDIO_MINIMIZE_LATENCY ) buffers = 2; + if ( buffers < 2 ) buffers = 3; + temp = ((int) buffers << 16) + (int)( log10( (double)ossBufferBytes ) / log10( 2.0 ) ); + result = ioctl( fd, SNDCTL_DSP_SETFRAGMENT, &temp ); + if ( result == -1 ) { + close( fd ); + errorStream_ << "RtApiOss::probeDeviceOpen: error setting buffer size on device (" << ainfo.name << ")."; + errorText_ = errorStream_.str(); + return FAILURE; + } + stream_.nBuffers = buffers; + + // Save buffer size (in sample frames). + *bufferSize = ossBufferBytes / ( formatBytes(stream_.deviceFormat[mode]) * deviceChannels ); + stream_.bufferSize = *bufferSize; + + // Set the sample rate. + int srate = sampleRate; + result = ioctl( fd, SNDCTL_DSP_SPEED, &srate ); + if ( result == -1 ) { + close( fd ); + errorStream_ << "RtApiOss::probeDeviceOpen: error setting sample rate (" << sampleRate << ") on device (" << ainfo.name << ")."; + errorText_ = errorStream_.str(); + return FAILURE; + } + + // Verify the sample rate setup worked. + if ( abs( srate - (int)sampleRate ) > 100 ) { + close( fd ); + errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") does not support sample rate (" << sampleRate << ")."; + errorText_ = errorStream_.str(); + return FAILURE; + } + stream_.sampleRate = sampleRate; + + if ( mode == INPUT && stream_.mode == OUTPUT && stream_.device[0] == device) { + // We're doing duplex setup here. + stream_.deviceFormat[0] = stream_.deviceFormat[1]; + stream_.nDeviceChannels[0] = deviceChannels; + } + + // Set interleaving parameters. + stream_.userInterleaved = true; + stream_.deviceInterleaved[mode] = true; + if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) + stream_.userInterleaved = false; + + // Set flags for buffer conversion + stream_.doConvertBuffer[mode] = false; + if ( stream_.userFormat != stream_.deviceFormat[mode] ) + stream_.doConvertBuffer[mode] = true; + if ( stream_.nUserChannels[mode] < stream_.nDeviceChannels[mode] ) + stream_.doConvertBuffer[mode] = true; + if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] && + stream_.nUserChannels[mode] > 1 ) + stream_.doConvertBuffer[mode] = true; + + // Allocate the stream handles if necessary and then save. + if ( stream_.apiHandle == 0 ) { + try { + handle = new OssHandle; + } + catch ( std::bad_alloc& ) { + errorText_ = "RtApiOss::probeDeviceOpen: error allocating OssHandle memory."; + goto error; + } + + if ( pthread_cond_init( &handle->runnable, NULL ) ) { + errorText_ = "RtApiOss::probeDeviceOpen: error initializing pthread condition variable."; + goto error; + } + + stream_.apiHandle = (void *) handle; + } + else { + handle = (OssHandle *) stream_.apiHandle; + } + handle->id[mode] = fd; + + // Allocate necessary internal buffers. + unsigned long bufferBytes; + bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat ); + stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 ); + if ( stream_.userBuffer[mode] == NULL ) { + errorText_ = "RtApiOss::probeDeviceOpen: error allocating user buffer memory."; + goto error; + } + + if ( stream_.doConvertBuffer[mode] ) { + + bool makeBuffer = true; + bufferBytes = stream_.nDeviceChannels[mode] * formatBytes( stream_.deviceFormat[mode] ); + if ( mode == INPUT ) { + if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) { + unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] ); + if ( bufferBytes <= bytesOut ) makeBuffer = false; + } + } + + if ( makeBuffer ) { + bufferBytes *= *bufferSize; + if ( stream_.deviceBuffer ) free( stream_.deviceBuffer ); + stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 ); + if ( stream_.deviceBuffer == NULL ) { + errorText_ = "RtApiOss::probeDeviceOpen: error allocating device buffer memory."; + goto error; + } + } + } + + stream_.device[mode] = device; + stream_.state = STREAM_STOPPED; + + // Setup the buffer conversion information structure. + if ( stream_.doConvertBuffer[mode] ) setConvertInfo( mode, firstChannel ); + + // Setup thread if necessary. + if ( stream_.mode == OUTPUT && mode == INPUT ) { + // We had already set up an output stream. + stream_.mode = DUPLEX; + if ( stream_.device[0] == device ) handle->id[0] = fd; + } + else { + stream_.mode = mode; + + // Setup callback thread. + stream_.callbackInfo.object = (void *) this; + + // Set the thread attributes for joinable and realtime scheduling + // priority. The higher priority will only take affect if the + // program is run as root or suid. + pthread_attr_t attr; + pthread_attr_init( &attr ); + pthread_attr_setdetachstate( &attr, PTHREAD_CREATE_JOINABLE ); +#ifdef SCHED_RR // Undefined with some OSes (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, ¶m); + } + 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( 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 ); + + #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 ); +} + +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; + +#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. +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 *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++) { + out[info.outOffset[j]] = (Int32) std::min(std::lround(in[info.inOffset[j]] * 2147483648.f), 2147483647L); + } + 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::min(std::lround(in[info.inOffset[j]] * 2147483648.0), 2147483647L); + } + 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::min(std::lround(in[info.inOffset[j]] * 8388608.f), 8388607L); + } + 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::min(std::lround(in[info.inOffset[j]] * 8388608.0), 8388607L); + } + 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::min(std::lround(in[info.inOffset[j]] * 32768.f), 32767L); + } + 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::min(std::lround(in[info.inOffset[j]] * 32768.0), 32767L); + } + 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::min(std::lround(in[info.inOffset[j]] * 128.f), 127L); + } + 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::min(std::lround(in[info.inOffset[j]] * 128.0), 127L); + } + 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 ) {
cbits/RtAudio.h view
@@ -5,12 +5,13 @@ 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 and ASIO) operating systems.+ (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) 2001-2012 Gary P. Scavone+ Copyright (c) 2001-2019 Gary P. Scavone Permission is hereby granted, free of charge, to any person obtaining a copy of this software and associated documentation files@@ -42,14 +43,29 @@ \file RtAudio.h */ -// RtAudio: Version 4.0.11- #ifndef __RTAUDIO_H #define __RTAUDIO_H +#define RTAUDIO_VERSION "5.1.0"++#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 "RtError.h"+#include <stdexcept>+#include <iostream> /*! \typedef typedef unsigned long RtAudioFormat; \brief RtAudio data format type.@@ -59,12 +75,10 @@ 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 24-bit data is expected to be encapsulated in a 32-bit- format. - \e RTAUDIO_SINT8: 8-bit signed integer. - \e RTAUDIO_SINT16: 16-bit signed integer.- - \e RTAUDIO_SINT24: Lower 3 bytes of 32-bit signed integer.+ - \e RTAUDIO_SINT24: 24-bit signed integer. - \e RTAUDIO_SINT32: 32-bit signed integer. - \e RTAUDIO_FLOAT32: Normalized between plus/minus 1.0. - \e RTAUDIO_FLOAT64: Normalized between plus/minus 1.0.@@ -72,7 +86,7 @@ typedef unsigned long RtAudioFormat; static const RtAudioFormat RTAUDIO_SINT8 = 0x1; // 8-bit signed integer. static const RtAudioFormat RTAUDIO_SINT16 = 0x2; // 16-bit signed integer.-static const RtAudioFormat RTAUDIO_SINT24 = 0x4; // Lower 3 bytes of 32-bit signed integer.+static const RtAudioFormat RTAUDIO_SINT24 = 0x4; // 24-bit signed integer. static const RtAudioFormat RTAUDIO_SINT32 = 0x8; // 32-bit signed integer. static const RtAudioFormat RTAUDIO_FLOAT32 = 0x10; // Normalized between plus/minus 1.0. static const RtAudioFormat RTAUDIO_FLOAT64 = 0x20; // Normalized between plus/minus 1.0.@@ -87,6 +101,7 @@ - \e RTAUDIO_MINIMIZE_LATENCY: Attempt to set stream parameters for lowest possible latency. - \e RTAUDIO_HOG_DEVICE: Attempt grab device for exclusive use. - \e RTAUDIO_ALSA_USE_DEFAULT: Use the "default" PCM device (ALSA only).+ - \e RTAUDIO_JACK_DONT_CONNECT: Do not automatically connect ports (JACK only). By default, RtAudio streams pass and receive audio data from the client in an interleaved format. By passing the@@ -102,7 +117,7 @@ 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@@ -118,6 +133,9 @@ 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).@@ -125,6 +143,7 @@ static const RtAudioStreamFlags RTAUDIO_HOG_DEVICE = 0x4; // Attempt grab device and prevent use by others. static const RtAudioStreamFlags RTAUDIO_SCHEDULE_REALTIME = 0x8; // Try to select realtime scheduling for callback thread. static const RtAudioStreamFlags RTAUDIO_ALSA_USE_DEFAULT = 0x10; // Use the "default" PCM device (ALSA only).+static const RtAudioStreamFlags RTAUDIO_JACK_DONT_CONNECT = 0x20; // Do not automatically connect ports (JACK only). /*! \typedef typedef unsigned long RtAudioStreamStatus; \brief RtAudio stream status (over- or underflow) flags.@@ -175,6 +194,7 @@ \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@@ -186,7 +206,61 @@ 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 RTAUDIO_DLL_PUBLIC RtAudioError : public std::runtime_error+{+ 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 occured 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 occured. */+ SYSTEM_ERROR, /*!< A system error occured. */+ THREAD_ERROR /*!< A thread error occured. */+ };++ //! The constructor.+ RtAudioError( const std::string& message,+ Type type = RtAudioError::UNSPECIFIED )+ : std::runtime_error(message), type_(type) {}++ //! Prints thrown error message to stderr.+ virtual void printMessage( void ) const+ { std::cerr << '\n' << what() << "\n\n"; }++ //! Returns the thrown error message type.+ virtual const Type& getType(void) const { return type_; }++ //! Returns the thrown error message string.+ virtual const std::string getMessage(void) const+ { return std::string(what()); }++ protected:+ Type type_;+};++//! RtAudio error callback function prototype.+/*!+ \param type Type of error.+ \param errorText Error description.+ */+typedef void (*RtAudioErrorCallback)( RtAudioError::Type type, const std::string &errorText );+ // **************************************************************** // // // RtAudio class declaration.@@ -202,7 +276,7 @@ class RtApi; -class RtAudio+class RTAUDIO_DLL_PUBLIC RtAudio { public: @@ -214,9 +288,11 @@ 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_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.@@ -229,12 +305,13 @@ bool isDefaultOutput; /*!< true if this is the default output device. */ bool isDefaultInput; /*!< 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, e.g. 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), nativeFormats(0) {}+ isDefaultOutput(false), isDefaultInput(false), preferredSampleRate(0), nativeFormats(0) {} }; //! The structure for specifying input or ouput stream parameters.@@ -273,7 +350,7 @@ 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@@ -316,34 +393,60 @@ : flags(0), numberOfBuffers(0), priority(0) {} }; + //! A static function to determine the current RtAudio version.+ static std::string getVersion( void );+ //! A static function to determine the available compiled audio APIs. /*! The values returned in the std::vector can be compared against 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 ); + //! 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.+ /*!+ A case insensitive comparison will check the specified name+ against the list of compiled APIs, and return the one which+ matches. On failure, the function returns UNSPECIFIED.+ */+ static RtAudio::Api getCompiledApiByName( const std::string &name );+ //! The class constructor. /*!- The constructor performs minor initialization tasks. No exceptions- can be thrown.+ The constructor performs minor initialization tasks. An exception+ can be thrown if no API support is compiled. 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). */- RtAudio( RtAudio::Api api=UNSPECIFIED ) throw();+ RtAudio( RtAudio::Api api=UNSPECIFIED ); //! 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. /*!@@ -351,13 +454,13 @@ is called, thus supporting devices connected \e after instantiation. If a system error occurs during processing, a warning will be issued. */- unsigned int getDeviceCount( void ) throw();+ unsigned int getDeviceCount( void ); //! Return an RtAudio::DeviceInfo structure for a specified device number. /*! Any device integer between 0 and getDeviceCount() - 1 is valid.- If an invalid argument is provided, an RtError (type = INVALID_USE)+ 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@@ -374,7 +477,7 @@ client's responsibility to verify that a device is available before attempting to open a stream. */- unsigned int getDefaultOutputDevice( void ) throw();+ unsigned int getDefaultOutputDevice( void ); //! A function that returns the index of the default input device. /*!@@ -384,13 +487,13 @@ client's responsibility to verify that a device is available before attempting to open a stream. */- unsigned int getDefaultInputDevice( void ) throw();+ unsigned int getDefaultInputDevice( void ); //! A public function for opening a stream with the specified parameters. /*!- An RtError (type = SYSTEM_ERROR) is thrown if a stream cannot be+ An RtAudioError (type = SYSTEM_ERROR) is thrown if a stream cannot be opened with the specified parameters or an error occurs during- processing. An RtError (type = INVALID_USE) is thrown if any+ processing. An RtAudioError (type = INVALID_USE) is thrown if any invalid device ID or channel number parameters are specified. \param outputParameters Specifies output stream parameters to use@@ -423,24 +526,26 @@ 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 occured. */ 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 );+ void *userData = NULL, RtAudio::StreamOptions *options = NULL, RtAudioErrorCallback errorCallback = 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). */- void closeStream( void ) throw();+ void closeStream( void ); //! A function that starts a stream. /*!- An RtError (type = SYSTEM_ERROR) is thrown if an error occurs- during processing. An RtError (type = INVALID_USE) is thrown if a+ 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. */@@ -448,8 +553,8 @@ //! Stop a stream, allowing any samples remaining in the output queue to be played. /*!- An RtError (type = SYSTEM_ERROR) is thrown if an error occurs- during processing. An RtError (type = INVALID_USE) is thrown if a+ 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. */@@ -457,32 +562,38 @@ //! Stop a stream, discarding any samples remaining in the input/output queue. /*!- An RtError (type = SYSTEM_ERROR) is thrown if an error occurs- during processing. An RtError (type = INVALID_USE) is thrown if a+ 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. */ void abortStream( void ); //! Returns true if a stream is open and false if not.- bool isStreamOpen( void ) const throw();+ bool isStreamOpen( void ) const; //! Returns true if the stream is running and false if it is stopped or not open.- bool isStreamRunning( void ) const throw();+ bool isStreamRunning( void ) const; //! Returns the number of elapsed seconds since the stream was started. /*!- If a stream is not open, an RtError (type = INVALID_USE) will be thrown.+ If a stream is not open, an RtAudioError (type = INVALID_USE) will be thrown. */ 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. /*! 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- RtError (type = INVALID_USE) will be thrown. If the API does not+ RtAudioError (type = INVALID_USE) will be thrown. If the API does not report latency, the return value will be zero. */ long getStreamLatency( void );@@ -491,12 +602,12 @@ /*! 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 RtError (type = INVALID_USE) will be thrown.+ open, an RtAudioError (type = INVALID_USE) will be thrown. */ unsigned int getStreamSampleRate( void ); //! Specify whether warning messages should be printed to stderr.- void showWarnings( bool value = true ) throw();+ void showWarnings( bool value = true ); protected: @@ -505,11 +616,16 @@ }; // Operating system dependent thread functionality.-#if defined(__WINDOWS_DS__) || defined(__WINDOWS_ASIO__)+#if defined(__WINDOWS_DS__) || defined(__WINDOWS_ASIO__) || defined(__WINDOWS_WASAPI__)++ #ifndef NOMINMAX+ #define NOMINMAX+ #endif #include <windows.h> #include <process.h>+ #include <stdint.h> - typedef unsigned long ThreadHandle;+ typedef uintptr_t ThreadHandle; typedef CRITICAL_SECTION StreamMutex; #elif defined(__LINUX_ALSA__) || defined(__LINUX_PULSE__) || defined(__UNIX_JACK__) || defined(__LINUX_OSS__) || defined(__MACOSX_CORE__)@@ -535,12 +651,15 @@ 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), apiInfo(0), isRunning(false) {}+ :object(0), callback(0), userData(0), errorCallback(0), apiInfo(0), isRunning(false), doRealtime(false), priority(0) {} }; // **************************************************************** //@@ -557,13 +676,42 @@ // // **************************************************************** // +#pragma pack(push, 1)+class S24 {++ protected:+ unsigned char c3[3];++ public:+ S24() {}++ S24& operator = ( const int& i ) {+ c3[0] = (i & 0x000000ff);+ c3[1] = (i & 0x0000ff00) >> 8;+ c3[2] = (i & 0x00ff0000) >> 16;+ return *this;+ }++ S24( const double& d ) { *this = (int) d; }+ S24( const float& f ) { *this = (int) f; }+ S24( const signed short& s ) { *this = (int) s; }+ S24( const char& c ) { *this = (int) c; }++ int asInt() {+ int i = c3[0] | (c3[1] << 8) | (c3[2] << 16);+ if (i & 0x800000) i |= ~0xffffff;+ return i;+ }+};+#pragma pack(pop)+ #if defined( HAVE_GETTIMEOFDAY ) #include <sys/time.h> #endif #include <sstream> -class RtApi+class RTAUDIO_DLL_PUBLIC RtApi { public: @@ -578,7 +726,8 @@ RtAudio::StreamParameters *inputParameters, RtAudioFormat format, unsigned int sampleRate, unsigned int *bufferFrames, RtAudioCallback callback,- void *userData, RtAudio::StreamOptions *options );+ void *userData, RtAudio::StreamOptions *options,+ RtAudioErrorCallback errorCallback ); virtual void closeStream( void ); virtual void startStream( void ) = 0; virtual void stopStream( void ) = 0;@@ -586,9 +735,10 @@ long getStreamLatency( void ); unsigned int getStreamSampleRate( void ); virtual double getStreamTime( void );- bool isStreamOpen( void ) const { return stream_.state != STREAM_CLOSED; };- bool isStreamRunning( void ) const { return stream_.state == STREAM_RUNNING; };- void showWarnings( bool value ) { showWarnings_ = value; };+ virtual void setStreamTime( double time );+ bool isStreamOpen( void ) const { return stream_.state != STREAM_CLOSED; }+ bool isStreamRunning( void ) const { return stream_.state == STREAM_RUNNING; }+ void showWarnings( bool value ) { showWarnings_ = value; } protected:@@ -655,6 +805,7 @@ :apiHandle(0), deviceBuffer(0) { device[0] = 11111; device[1] = 11111; } }; + typedef S24 Int24; typedef signed short Int16; typedef signed int Int32; typedef float Float32;@@ -664,6 +815,7 @@ std::string errorText_; bool showWarnings_; RtApiStream stream_;+ bool firstErrorOccurred_; /*! Protected, api-specific method that attempts to open a device@@ -684,13 +836,13 @@ void clearStreamInfo(); /*!- Protected common method that throws an RtError (type =+ 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( RtError::Type type );+ void error( RtAudioError::Type type ); /*! Protected method used to perform format, channel number, and/or interleaving@@ -714,21 +866,22 @@ // // **************************************************************** // -inline RtAudio::Api RtAudio :: getCurrentApi( void ) throw() { return rtapi_->getCurrentApi(); }-inline unsigned int RtAudio :: getDeviceCount( void ) throw() { return rtapi_->getDeviceCount(); }+inline RtAudio::Api RtAudio :: getCurrentApi( void ) { return rtapi_->getCurrentApi(); }+inline unsigned int RtAudio :: getDeviceCount( void ) { return rtapi_->getDeviceCount(); } inline RtAudio::DeviceInfo RtAudio :: getDeviceInfo( unsigned int device ) { return rtapi_->getDeviceInfo( device ); }-inline unsigned int RtAudio :: getDefaultInputDevice( void ) throw() { return rtapi_->getDefaultInputDevice(); }-inline unsigned int RtAudio :: getDefaultOutputDevice( void ) throw() { return rtapi_->getDefaultOutputDevice(); }-inline void RtAudio :: closeStream( void ) throw() { return rtapi_->closeStream(); }+inline unsigned int RtAudio :: getDefaultInputDevice( void ) { return rtapi_->getDefaultInputDevice(); }+inline unsigned int RtAudio :: getDefaultOutputDevice( void ) { return rtapi_->getDefaultOutputDevice(); }+inline void RtAudio :: closeStream( void ) { return rtapi_->closeStream(); } inline void RtAudio :: startStream( void ) { return rtapi_->startStream(); } inline void RtAudio :: stopStream( void ) { return rtapi_->stopStream(); } inline void RtAudio :: abortStream( void ) { return rtapi_->abortStream(); }-inline bool RtAudio :: isStreamOpen( void ) const throw() { return rtapi_->isStreamOpen(); }-inline bool RtAudio :: isStreamRunning( void ) const throw() { return rtapi_->isStreamRunning(); }+inline bool RtAudio :: isStreamOpen( void ) const { return rtapi_->isStreamOpen(); }+inline bool RtAudio :: isStreamRunning( void ) const { return rtapi_->isStreamRunning(); } inline long RtAudio :: getStreamLatency( void ) { return rtapi_->getStreamLatency(); }-inline unsigned int RtAudio :: getStreamSampleRate( void ) { return rtapi_->getStreamSampleRate(); };+inline unsigned int RtAudio :: getStreamSampleRate( void ) { return rtapi_->getStreamSampleRate(); } inline double RtAudio :: getStreamTime( void ) { return rtapi_->getStreamTime(); }-inline void RtAudio :: showWarnings( bool value ) throw() { rtapi_->showWarnings( value ); }+inline void RtAudio :: setStreamTime( double time ) { return rtapi_->setStreamTime( time ); }+inline void RtAudio :: showWarnings( bool value ) { rtapi_->showWarnings( value ); } // RtApi Subclass prototypes. @@ -742,7 +895,7 @@ RtApiCore(); ~RtApiCore();- RtAudio::Api getCurrentApi( void ) { return RtAudio::MACOSX_CORE; };+ RtAudio::Api getCurrentApi( void ) { return RtAudio::MACOSX_CORE; } unsigned int getDeviceCount( void ); RtAudio::DeviceInfo getDeviceInfo( unsigned int device ); unsigned int getDefaultOutputDevice( void );@@ -751,7 +904,6 @@ 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,@@ -780,14 +932,13 @@ RtApiJack(); ~RtApiJack();- RtAudio::Api getCurrentApi( void ) { return RtAudio::UNIX_JACK; };+ 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,@@ -801,6 +952,8 @@ unsigned int firstChannel, unsigned int sampleRate, RtAudioFormat format, unsigned int *bufferSize, RtAudio::StreamOptions *options );++ bool shouldAutoconnect_; }; #endif@@ -813,14 +966,13 @@ RtApiAsio(); ~RtApiAsio();- RtAudio::Api getCurrentApi( void ) { return RtAudio::WINDOWS_ASIO; };+ 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,@@ -849,7 +1001,7 @@ RtApiDs(); ~RtApiDs();- RtAudio::Api getCurrentApi( void ) { return RtAudio::WINDOWS_DS; };+ RtAudio::Api getCurrentApi( void ) { return RtAudio::WINDOWS_DS; } unsigned int getDeviceCount( void ); unsigned int getDefaultOutputDevice( void ); unsigned int getDefaultInputDevice( void );@@ -858,7 +1010,6 @@ 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,@@ -871,6 +1022,7 @@ 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,@@ -879,6 +1031,43 @@ #endif +#if defined(__WINDOWS_WASAPI__)++struct IMMDeviceEnumerator;++class RtApiWasapi : public RtApi+{+public:+ RtApiWasapi();+ virtual ~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@@ -887,7 +1076,7 @@ RtApiAlsa(); ~RtApiAlsa();- RtAudio::Api getCurrentApi() { return RtAudio::LINUX_ALSA; };+ RtAudio::Api getCurrentApi() { return RtAudio::LINUX_ALSA; } unsigned int getDeviceCount( void ); RtAudio::DeviceInfo getDeviceInfo( unsigned int device ); void closeStream( void );@@ -919,7 +1108,7 @@ { public: ~RtApiPulse();- RtAudio::Api getCurrentApi() { return RtAudio::LINUX_PULSE; };+ RtAudio::Api getCurrentApi() { return RtAudio::LINUX_PULSE; } unsigned int getDeviceCount( void ); RtAudio::DeviceInfo getDeviceInfo( unsigned int device ); void closeStream( void );@@ -935,8 +1124,7 @@ private: - std::vector<RtAudio::DeviceInfo> devices_;- void saveDeviceInfo( void );+ void collectDeviceInfo( void ); bool probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels, unsigned int firstChannel, unsigned int sampleRate, RtAudioFormat format, unsigned int *bufferSize,@@ -953,7 +1141,7 @@ RtApiOss(); ~RtApiOss();- RtAudio::Api getCurrentApi() { return RtAudio::LINUX_OSS; };+ RtAudio::Api getCurrentApi() { return RtAudio::LINUX_OSS; } unsigned int getDeviceCount( void ); RtAudio::DeviceInfo getDeviceInfo( unsigned int device ); void closeStream( void );@@ -983,21 +1171,21 @@ { public: - RtApiDummy() { errorText_ = "RtApiDummy: This class provides no functionality."; error( RtError::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 ) {};+ 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; };+ 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
− cbits/RtError.h
@@ -1,60 +0,0 @@-/************************************************************************/-/*! \class RtError- \brief Exception handling class for RtAudio & RtMidi.-- The RtError class is quite simple but it does allow errors to be- "caught" by RtError::Type. See the RtAudio and RtMidi- documentation to know which methods can throw an RtError.--*/-/************************************************************************/--#ifndef RTERROR_H-#define RTERROR_H--#include <exception>-#include <iostream>-#include <string>--class RtError : public std::exception-{- public:- //! Defined RtError 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 occured 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 occured. */- SYSTEM_ERROR, /*!< A system error occured. */- THREAD_ERROR /*!< A thread error occured. */- };-- //! The constructor.- RtError( const std::string& message, Type type = RtError::UNSPECIFIED ) throw() : message_(message), type_(type) {}- - //! The destructor.- virtual ~RtError( 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_;-};--#endif
cbits/proAudio.cpp view
@@ -8,108 +8,108 @@ //--- class AudioSample -------------------------------------------- AudioSample::AudioSample(const AudioSample & source) :- m_size(source.m_size), m_channels(source.m_channels), m_sampleRate(source.m_sampleRate), m_bitsPerSample(source.m_bitsPerSample) { - m_data = new unsigned char [m_size]; memcpy(m_data,source.m_data, m_size); + m_size(source.m_size), m_channels(source.m_channels), m_sampleRate(source.m_sampleRate), m_bitsPerSample(source.m_bitsPerSample) {+ m_data = new unsigned char [m_size]; memcpy(m_data,source.m_data, m_size); } bool AudioSample::bitsPerSample(unsigned short bits) {- if(bits==16) {- if(m_bitsPerSample==8) {- unsigned char* data = new unsigned char[2*m_size];- for(unsigned int i=0; i<m_size; ++i) {- signed short *ptr =(signed short*)data+i;- *ptr = m_data[i]*255;- }- delete [] m_data;- m_data = data;- m_size*=2;- return true;- }- else if(m_bitsPerSample==16) {- return true; // nothing to do- }- else if(m_bitsPerSample==32) { // float, normalized from -1.0f to 1.0f- unsigned char* data = new unsigned char[m_size/2];- for(unsigned int i=0; i<m_size/4; ++i) {- signed short *ptr =(signed short*)data+i;- float* src=(float*)m_data+i;- *ptr =(signed short)(*src*SHRT_MAX);- }- delete [] m_data;- m_data = data;- m_size/=2;- return true;- }- }- fprintf(stderr,"AudioSample::bitsPerSample ERROR: conversion from %i to %i bits not supported.\n", m_bitsPerSample, bits);- return false;+ if(bits==16) {+ if(m_bitsPerSample==8) {+ unsigned char* data = new unsigned char[2*m_size];+ for(unsigned int i=0; i<m_size; ++i) {+ signed short *ptr =(signed short*)data+i;+ *ptr = m_data[i]*255;+ }+ delete [] m_data;+ m_data = data;+ m_size*=2;+ return true;+ }+ else if(m_bitsPerSample==16) {+ return true; // nothing to do+ }+ else if(m_bitsPerSample==32) { // float, normalized from -1.0f to 1.0f+ unsigned char* data = new unsigned char[m_size/2];+ for(unsigned int i=0; i<m_size/4; ++i) {+ signed short *ptr =(signed short*)data+i;+ float* src=(float*)m_data+i;+ *ptr =(signed short)(*src*SHRT_MAX);+ }+ delete [] m_data;+ m_data = data;+ m_size/=2;+ return true;+ }+ }+ fprintf(stderr,"AudioSample::bitsPerSample ERROR: conversion from %i to %i bits not supported.\n", m_bitsPerSample, bits);+ return false; } void AudioSample::volume(float f) {- if(m_bitsPerSample==8) for(signed char *ptr =(signed char *)m_data; ptr<(signed char *)m_data+m_size; ++ptr) {- float value=*ptr * f;- if(value>CHAR_MAX) *ptr =CHAR_MAX;- else if(value<CHAR_MIN) *ptr =CHAR_MIN;- else *ptr =(signed char)value;- }- else if(m_bitsPerSample==16) for(signed short *ptr =(signed short*)m_data; ptr<(signed short*)m_data+m_size/2; ++ptr) {- float value=*ptr * f;- if(value>SHRT_MAX) *ptr =SHRT_MAX;- else if(value<SHRT_MIN) *ptr =SHRT_MIN;- else *ptr =(signed short)value;- }- else if(m_bitsPerSample==32) for(float *ptr =(float*)m_data; ptr<(float*)m_data+m_size/4; ++ptr) {- *ptr *= f;- if(*ptr>1.0f) *ptr=1.0f; - else if(*ptr<-1.0f) *ptr=-1.0f;- }- else fprintf(stderr,"AudioSample::changeVolume ERROR: %i bits per sample not supported.\n",m_bitsPerSample);+ if(m_bitsPerSample==8) for(signed char *ptr =(signed char *)m_data; ptr<(signed char *)m_data+m_size; ++ptr) {+ float value=*ptr * f;+ if(value>CHAR_MAX) *ptr =CHAR_MAX;+ else if(value<CHAR_MIN) *ptr =CHAR_MIN;+ else *ptr =(signed char)value;+ }+ else if(m_bitsPerSample==16) for(signed short *ptr =(signed short*)m_data; ptr<(signed short*)m_data+m_size/2; ++ptr) {+ float value=*ptr * f;+ if(value>SHRT_MAX) *ptr =SHRT_MAX;+ else if(value<SHRT_MIN) *ptr =SHRT_MIN;+ else *ptr =(signed short)value;+ }+ else if(m_bitsPerSample==32) for(float *ptr =(float*)m_data; ptr<(float*)m_data+m_size/4; ++ptr) {+ *ptr *= f;+ if(*ptr>1.0f) *ptr=1.0f;+ else if(*ptr<-1.0f) *ptr=-1.0f;+ }+ else fprintf(stderr,"AudioSample::changeVolume ERROR: %i bits per sample not supported.\n",m_bitsPerSample); } AudioSample* AudioSample::readWav(FILE* stream, size_t (*readFunc)( void *, size_t, size_t, FILE *)) {- char id[4]; //four unsigned chars to hold chunk IDs- readFunc(id,sizeof(unsigned char),4,stream);- if (strncmp(id,"RIFF",4)!=0) return 0;+ char id[4]; //four unsigned chars to hold chunk IDs+ readFunc(id,sizeof(unsigned char),4,stream);+ if (strncmp(id,"RIFF",4)!=0) return 0; - unsigned int size;- readFunc(&size,sizeof(unsigned int),1,stream);+ unsigned int size;+ readFunc(&size,sizeof(unsigned int),1,stream); - readFunc(id,sizeof(unsigned char),4,stream);- if (strncmp(id,"WAVE",4)!=0) return 0;+ readFunc(id,sizeof(unsigned char),4,stream);+ if (strncmp(id,"WAVE",4)!=0) return 0; - unsigned short encoding, block_align, channels, bitsPerSample;- unsigned int chunk_length, byte_rate, sampleRate;+ unsigned short encoding, block_align, channels, bitsPerSample;+ unsigned int chunk_length, byte_rate, sampleRate; - readFunc(id, sizeof(unsigned char), 4, stream); //read ID 'fmt ';- readFunc(&chunk_length, sizeof(unsigned int),1,stream); // header length, 16 expected - readFunc(&encoding, sizeof(short), 1, stream); // should be "1" for simple PCM data- if(encoding!=1) return 0;+ readFunc(id, sizeof(unsigned char), 4, stream); //read ID 'fmt ';+ readFunc(&chunk_length, sizeof(unsigned int),1,stream); // header length, 16 expected+ readFunc(&encoding, sizeof(short), 1, stream); // should be "1" for simple PCM data+ if(encoding!=1) return 0; - readFunc(&channels, sizeof(short),1,stream);- readFunc(&sampleRate, sizeof(unsigned int), 1, stream);- readFunc(&byte_rate, sizeof(unsigned int), 1, stream);- readFunc(&block_align, sizeof(short), 1, stream);- readFunc(&bitsPerSample, sizeof(short), 1, stream);- - readFunc(id, sizeof(unsigned char), 4, stream); // read ID 'data'- readFunc(&size, sizeof(unsigned int), 1, stream);- unsigned char *data = new unsigned char[size];- readFunc(data, sizeof(unsigned char), size, stream);- - return new AudioSample(data,size, channels, sampleRate, bitsPerSample);+ readFunc(&channels, sizeof(short),1,stream);+ readFunc(&sampleRate, sizeof(unsigned int), 1, stream);+ readFunc(&byte_rate, sizeof(unsigned int), 1, stream);+ readFunc(&block_align, sizeof(short), 1, stream);+ readFunc(&bitsPerSample, sizeof(short), 1, stream);++ readFunc(id, sizeof(unsigned char), 4, stream); // read ID 'data'+ readFunc(&size, sizeof(unsigned int), 1, stream);+ unsigned char *data = new unsigned char[size];+ readFunc(data, sizeof(unsigned char), size, stream);++ return new AudioSample(data,size, channels, sampleRate, bitsPerSample); } AudioSample* AudioSample::loadWav(const std::string & fname) { #ifdef _MSC_VER- FILE *fp = 0;- fopen_s(&fp, fname.c_str(), "rb");+ FILE *fp = 0;+ fopen_s(&fp, fname.c_str(), "rb"); #else- FILE *fp = fopen(fname.c_str(), "rb");+ FILE *fp = fopen(fname.c_str(), "rb"); #endif- if (!fp) return 0;- AudioSample * pSample = readWav(fp, fread);- fclose(fp);- return pSample;+ if (!fp) return 0;+ AudioSample * pSample = readWav(fp, fread);+ fclose(fp);+ return pSample; } //--- class DeviceAudio --------------------------------------------@@ -121,17 +121,17 @@ }; static AudioSample* loadOgg(const std::string & fname) {- int channels, sampleRate;- short *decoded;- int len = stb_vorbis_decode_filename(const_cast<char*>(fname.c_str()), &channels, &sampleRate, &decoded);- if(len<0) return 0;- // convert to AudioSample:- unsigned int size = len*channels*sizeof(short);- unsigned char * data = new unsigned char[size];- if(!data) return 0;- memcpy(data,decoded, size);- free(decoded);- return new AudioSample(data, size, channels, sampleRate, 16);+ int channels, sampleRate;+ short *decoded;+ int len = stb_vorbis_decode_filename(const_cast<char*>(fname.c_str()), &channels, &sampleRate, &decoded);+ if(len<0) return 0;+ // convert to AudioSample:+ unsigned int size = len*channels*sizeof(short);+ unsigned char * data = new unsigned char[size];+ if(!data) return 0;+ memcpy(data,decoded, size);+ free(decoded);+ return new AudioSample(data, size, channels, sampleRate, 16); } static string toLower(const string & s) {@@ -142,26 +142,26 @@ } DeviceAudio::DeviceAudio() : m_freqOut(0), m_volL(1.0f), m_volR(1.0f) {- loaderRegister(AudioSample::loadWav,"wav");- loaderRegister(loadOgg,"ogg");+ loaderRegister(AudioSample::loadWav,"wav");+ loaderRegister(loadOgg,"ogg"); } -unsigned int DeviceAudio::sampleFromFile(const std::string & filename, float volume) { - if(filename.rfind('.')>filename.size()) return 0;- string suffix=toLower(filename.substr(filename.rfind('.')+1));- map<string, AudioSample * (*)(const string &)>::iterator it = mm_loader.find(suffix);- if(it==mm_loader.end()) return 0;- AudioSample* pSample = (*(it->second))(filename);- if(!pSample) return 0;- unsigned int ret = sampleFromMemory(*pSample, volume);- delete pSample;- return ret; +uint64_t DeviceAudio::sampleFromFile(const std::string & filename, float volume) {+ if(filename.rfind('.')>filename.size()) return 0;+ string suffix=toLower(filename.substr(filename.rfind('.')+1));+ map<string, AudioSample * (*)(const string &)>::iterator it = mm_loader.find(suffix);+ if(it==mm_loader.end()) return 0;+ AudioSample* pSample = (*(it->second))(filename);+ if(!pSample) return 0;+ uint64_t ret = sampleFromMemory(*pSample, volume);+ delete pSample;+ return ret; } bool DeviceAudio::loaderRegister(AudioSample *(*loadFunc)(const std::string &), const std::string & suffix) {- return mm_loader.insert(std::make_pair(toLower(suffix),loadFunc)).second;+ return mm_loader.insert(std::make_pair(toLower(suffix),loadFunc)).second; } bool DeviceAudio::loaderAvailable(const std::string & suffix) const {- return mm_loader.find(toLower(suffix))!=mm_loader.end();+ return mm_loader.find(toLower(suffix))!=mm_loader.end(); }
cbits/proAudio.h view
@@ -1,17 +1,39 @@ #ifndef _PRO_AUDIO #define _PRO_AUDIO +#include <cstdint>+/*+ Unique Handle (uint64) =+ UniqueId (HI bits: Global Monotonic Counter) + Payload (LO bits: i.e. sample index)+*/++#define UH_PAYLOAD_BIT_WIDTH 10+#define UH_PAYLOAD_MASK ((1 << UH_PAYLOAD_BIT_WIDTH) - 1)++// Unique Handle API+inline uint64_t UH_PACK_UNIQUE_HANDLE(uint64_t uid, uint16_t data) {+ return (uid << UH_PAYLOAD_BIT_WIDTH) | (data & UH_PAYLOAD_MASK);+}++inline uint64_t UH_UNPACK_UNIQUE_ID(uint64_t uh) {+ return uh >> UH_PAYLOAD_BIT_WIDTH;+}++inline uint16_t UH_UNPACK_PAYLOAD(uint64_t uh) {+ return uh & UH_PAYLOAD_MASK;+}+ #include <string> #include <map> /** @file proAudio.h \brief Public interface of proteaAudio- + Contains the declaration of the audio sample class and the abstract base class for audio mixer/playback devices- + \author Gerald Franz, www.viremo.de \version 0.6- + License notice (zlib license): (c) 2009 by Gerald Franz, www.viremo.de@@ -37,53 +59,53 @@ /// class representing an audio sample class AudioSample { public:- /// constructor from memory data- AudioSample(unsigned char * data, unsigned int size, unsigned short channels, unsigned int sampleRate, unsigned short bitsPerSample) :- m_data(data), m_size(size), m_channels(channels), m_sampleRate(sampleRate), m_bitsPerSample(bitsPerSample) { }- /// copy constructor- AudioSample(const AudioSample & source);- /// destructor- ~AudioSample() { delete[] m_data; }- - /// allows accessing sample data- unsigned char * data() { return m_data; };- /// allows reading sample data- const unsigned char * data() const { return m_data; };- /// returns sample size in bytes- unsigned int size() const { return m_size; }- /// returns sample size in number of frames- unsigned int frames() const { return m_size/m_channels/(m_bitsPerSample>>3); }- /// returns size of a single frame in bytes- unsigned int sizeFrame() const { return m_channels*(m_bitsPerSample>>3); }- /// returns number of parallel channels, 1 mono, 2 stereo- unsigned short channels() const { return m_channels; }- /// returns number of frames per second, e.g., 44100, 22050- unsigned int sampleRate() const { return m_sampleRate; }- /// returns number of bits per mono sample, e.g., 8, 16- unsigned short bitsPerSample() const { return m_bitsPerSample; }- /// converts to a different bit rate, e.g., 8, 16- bool bitsPerSample(unsigned short bits);- /// returns number of bytes per sample, e.g., 1, 2- unsigned short bytesPerSample() const { return m_bitsPerSample>>3; }+ /// constructor from memory data+ AudioSample(unsigned char * data, unsigned int size, unsigned short channels, unsigned int sampleRate, unsigned short bitsPerSample) :+ m_data(data), m_size(size), m_channels(channels), m_sampleRate(sampleRate), m_bitsPerSample(bitsPerSample) { }+ /// copy constructor+ AudioSample(const AudioSample & source);+ /// destructor+ ~AudioSample() { delete[] m_data; } - /// changes volume by given factor- void volume(float f);- - /// loads a WAV file- static AudioSample* loadWav(const std::string & fname);- /// reads WAV data from a stream via a function compatible to std::fread- static AudioSample* readWav(FILE* stream, size_t (*readFunc)( void *, size_t, size_t, FILE *));+ /// allows accessing sample data+ unsigned char * data() { return m_data; };+ /// allows reading sample data+ const unsigned char * data() const { return m_data; };+ /// returns sample size in bytes+ unsigned int size() const { return m_size; }+ /// returns sample size in number of frames+ unsigned int frames() const { return m_size/m_channels/(m_bitsPerSample>>3); }+ /// returns size of a single frame in bytes+ unsigned int sizeFrame() const { return m_channels*(m_bitsPerSample>>3); }+ /// returns number of parallel channels, 1 mono, 2 stereo+ unsigned short channels() const { return m_channels; }+ /// returns number of frames per second, e.g., 44100, 22050+ unsigned int sampleRate() const { return m_sampleRate; }+ /// returns number of bits per mono sample, e.g., 8, 16+ unsigned short bitsPerSample() const { return m_bitsPerSample; }+ /// converts to a different bit rate, e.g., 8, 16+ bool bitsPerSample(unsigned short bits);+ /// returns number of bytes per sample, e.g., 1, 2+ unsigned short bytesPerSample() const { return m_bitsPerSample>>3; }++ /// changes volume by given factor+ void volume(float f);++ /// loads a WAV file+ static AudioSample* loadWav(const std::string & fname);+ /// reads WAV data from a stream via a function compatible to std::fread+ static AudioSample* readWav(FILE* stream, size_t (*readFunc)( void *, size_t, size_t, FILE *)); protected:- /// stores sample data- unsigned char * m_data;- /// sample size in bytes- unsigned int m_size;- /// number of parallel channels, 1 mono, 2 stereo- unsigned short m_channels;- /// number of samples per second, e.g., 44100, 22050- unsigned int m_sampleRate;- /// number of bits per sample, e.g., 8, 16- unsigned short m_bitsPerSample;+ /// stores sample data+ unsigned char * m_data;+ /// sample size in bytes+ unsigned int m_size;+ /// number of parallel channels, 1 mono, 2 stereo+ unsigned short m_channels;+ /// number of samples per second, e.g., 44100, 22050+ unsigned int m_sampleRate;+ /// number of bits per sample, e.g., 8, 16+ unsigned short m_bitsPerSample; }; //--- class DeviceAudio --------------------------------------------@@ -92,30 +114,30 @@ class DeviceAudio { public: /// returns singleton object- /** This call is only allowed after a successful precedent creation of an audio device */+ /** This call is only allowed after a successful precedent creation of an audio device */ static DeviceAudio& singleton() { return *s_instance; } /// calls the destructor of the singleton object static void destroy() { if(s_instance) delete s_instance; s_instance=0; }; - /// sets master volume+ /// sets master volume void volume(float left, float right) { m_volL=left; m_volR=right; } /// sets master volume void volume(float leftAndRight) { m_volL=m_volR=leftAndRight; }- /// registers an audio sample loader function handling a file type identified by suffix- /** The function has to be of type AudioSample * loadXYZ(const std::string & filename).*/- bool loaderRegister(AudioSample *(*loadFunc)(const std::string &), const std::string & suffix);- /// returns true in case a loader for this file type is available- bool loaderAvailable(const std::string & suffix) const;+ /// registers an audio sample loader function handling a file type identified by suffix+ /** The function has to be of type AudioSample * loadXYZ(const std::string & filename).*/+ bool loaderRegister(AudioSample *(*loadFunc)(const std::string &), const std::string & suffix);+ /// returns true in case a loader for this file type is available+ bool loaderAvailable(const std::string & suffix) const; /// loads a sound sample from file, optionally adjusts volume, returns handle- virtual unsigned int sampleFromFile(const std::string & filename, float volume=1.0f);+ virtual uint64_t sampleFromFile(const std::string & filename, float volume=1.0f); /// converts a sound sample to internal audio format, returns handle- virtual unsigned int sampleFromMemory(const AudioSample & sample, float volume=1.0f)=0;- /// deletes a previously created sound sample resource identified by its handle- virtual bool sampleDestroy(unsigned int sample)=0;- /// allows read access to a sample identified by its handle- virtual const AudioSample* sample(unsigned int handle) const { return 0; }- + virtual uint64_t sampleFromMemory(const AudioSample & sample, float volume=1.0f)=0;+ /// deletes a previously created sound sample resource identified by its handle+ virtual bool sampleDestroy(uint64_t sample)=0;+ /// allows read access to a sample identified by its handle+ virtual const AudioSample* sample(uint64_t handle) const { return 0; }+ /// plays a specified sound sample once and sets its parameters /** \param sample handle of a previously loaded sample \param volumeL (optional) left volume@@ -123,7 +145,7 @@ \param disparity (optional) time difference between left and right channel in seconds. Use negative values to specify a delay for the left channel, positive for the right. \param pitch (optional) pitch factor for playback. 0.5 corresponds to one octave below, 2.0 to one above the original sample. \return a handle to the currently played sound or -1 in case of error */- virtual unsigned int soundPlay(unsigned int sample, float volumeL=1.0f, float volumeR=1.0f, float disparity=0.0f, float pitch=1.0f )=0;+ virtual uint64_t soundPlay(uint64_t sample, float volumeL=1.0f, float volumeR=1.0f, float disparity=0.0f, float pitch=1.0f )=0; /// plays a specified sound sample continuously and sets its parameters /** \param sample handle of a previously loaded sample \param volumeL (optional) left volume@@ -131,7 +153,25 @@ \param disparity (optional) time difference between left and right channel in seconds. Use negative values to specify a delay for the left channel, positive for the right. \param pitch (optional) pitch factor for playback. 0.5 corresponds to one octave below, 2.0 to one above the original sample. \return a handle to the currently played sound or -1 in case of error */- virtual unsigned int soundLoop(unsigned int sample, float volumeL=1.0f, float volumeR=1.0f, float disparity=0.0f, float pitch=1.0f )=0;+ virtual uint64_t soundLoop(uint64_t sample, float volumeL=1.0f, float volumeR=1.0f, float disparity=0.0f, float pitch=1.0f )=0;+ /// plays a specified sample once on a specific channel and sets its parameters+ /** \param i a channel number with the first channel starting at 0 + \param sample a sample handle returned by a previous load() call+ \param volumeL (optional) left volume+ \param volumeR (optional) right volume+ \param disparity (optional) time difference between left and right channel in seconds. Use negative values to specify a delay for the left channel, positive for the right.+ \param pitch (optional) pitch factor for playback. 0.5 corresponds to one octave below, 2.0 to one above the original sample.+ \return a handle to the currently played sound or 0 in case of error */+ virtual uint64_t soundPlayOn(unsigned int i, uint64_t sample, float volumeL=1.0f, float volumeR=1.0f, float disparity=0.0f, float pitch=1.0f )=0;+ /// plays a specified sample continuously on a specific channel and sets its parameters+ /** \param i a channel number with the first channel starting at 0 + \param sample a sample handle returned by a previous load() call+ \param volumeL (optional) left volume+ \param volumeR (optional) right volume+ \param disparity (optional) time difference between left and right channel in seconds. Use negative values to specify a delay for the left channel, positive for the right.+ \param pitch (optional) pitch factor for playback. 0.5 corresponds to one octave below, 2.0 to one above the original sample.+ \return a handle to the currently played sound or 0 in case of error */+ virtual uint64_t soundLoopOn(unsigned int i, uint64_t sample, float volumeL=1.0f, float volumeR=1.0f, float disparity=0.0f, float pitch=1.0f )=0; /// updates parameters of a specified sound /** \param sound handle of a currently active sound \param volumeL left volume@@ -139,29 +179,31 @@ \param disparity (optional) time difference between left and right channel in seconds. Use negative values to specify a delay for the left channel, positive for the right. \param pitch (optional) pitch factor for playback. 0.5 corresponds to one octave below, 2.0 to one above the original sample. \return true in case the parameters have been updated successfully */- virtual bool soundUpdate(unsigned int sound, float volumeL, float volumeR, float disparity=0.0f, float pitch=1.0f )=0;+ virtual bool soundUpdate(uint64_t sound, float volumeL, float volumeR, float disparity=0.0f, float pitch=1.0f )=0; /// stops a specified sound immediately- virtual bool soundStop(unsigned int sound)=0;+ virtual bool soundStop(uint64_t sound)=0; /// stops all sounds immediately virtual void soundStop()=0;- /// returns number of currently active sounds- virtual unsigned int soundActive() const=0;+ /// returns number of currently active sounds+ virtual unsigned int soundActive() const=0;+ /// check if a specified sound is active+ virtual bool soundActive(uint64_t sound)=0; protected:- /// constructor- DeviceAudio();+ /// constructor+ DeviceAudio(); /// destructor virtual ~DeviceAudio() { s_instance = 0; }- - /// stores output stream frequency- unsigned int m_freqOut;++ /// stores output stream frequency+ unsigned int m_freqOut; /// stores left master volume float m_volL; /// stores right master volume float m_volR;- /// map associating suffixes to loader functions- std::map<std::string, AudioSample * (*)(const std::string &)> mm_loader;- + /// map associating suffixes to loader functions+ std::map<std::string, AudioSample * (*)(const std::string &)> mm_loader;+ /// pointer to singleton static DeviceAudio * s_instance; };
cbits/proAudioRt.cpp view
@@ -5,14 +5,18 @@ #include <cstring> #include <cstdlib> +#ifdef __RTAUDIO_DUMMY__+#error "No suitable audio backend has found for RtAudio! Install the dev package with headers for the system's audio library and make sure it is in the include path!"+#endif+ using namespace std; struct _AudioTrack {- /// sample- AudioSample * sample;- - /// position in sample in frames- unsigned int dpos;+ /// sample+ AudioSample * sample;++ /// position in sample in frames+ unsigned int dpos; /// length of sample in frames unsigned int dlen; /// disparity in seconds between left and right, normally 0.0f@@ -27,96 +31,102 @@ bool isLoop; /// stores whether sample is currently playing bool isPlaying;+ /// unique id for this sound+ uint64_t uniqueId; }; DeviceAudio* DeviceAudioRt::create(unsigned int nTracks, unsigned int frequency, unsigned int chunkSize) { if(!s_instance) {- DeviceAudioRt* pAudio = new DeviceAudioRt(nTracks,frequency,chunkSize);- if(!pAudio->m_freqOut) delete pAudio;- else s_instance = pAudio;- }+ DeviceAudioRt* pAudio = new DeviceAudioRt(nTracks,frequency,chunkSize);+ if(!pAudio->m_freqOut) delete pAudio;+ else s_instance = pAudio;+ } return s_instance; } -DeviceAudioRt::DeviceAudioRt(unsigned int nTracks, unsigned int frequency, unsigned int chunkSize) : DeviceAudio() {- if ( m_dac.getDeviceCount() < 1 ) {- fprintf(stderr,"DeviceAudioRt ERROR: No audio devices found!\n");- return;- }- // Set our stream parameters for output only.- RtAudio::StreamParameters oParams;- oParams.deviceId = m_dac.getDefaultOutputDevice(); // default device- oParams.nChannels = 2; // stereo- oParams.firstChannel = 0;+DeviceAudioRt::DeviceAudioRt(unsigned int nTracks, unsigned int frequency, unsigned int chunkSize) : DeviceAudio(), m_uniqueCounter(0) {+ if ( m_dac.getDeviceCount() < 1 ) {+ fprintf(stderr,"DeviceAudioRt ERROR: No audio devices found!\n");+ return;+ }+ // Set our stream parameters for output only.+ RtAudio::StreamParameters oParams;+ oParams.deviceId = m_dac.getDefaultOutputDevice(); // default device+ oParams.nChannels = 2; // stereo+ oParams.firstChannel = 0; // initialize tracks: m_nSound=nTracks; ma_sound=new _AudioTrack[m_nSound];- memset(ma_sound,0,m_nSound*sizeof(_AudioTrack));- m_freqOut = frequency;+ memset(ma_sound,0,m_nSound*sizeof(_AudioTrack));+ m_freqOut = frequency; - try {- m_dac.openStream( &oParams, NULL, RTAUDIO_SINT16, frequency, &chunkSize, &cbMix, (void *)this );- m_dac.startStream();- }- catch ( RtError& e ) {- fprintf(stderr,"%s\n", e.getMessage().c_str());- if(m_dac.isStreamOpen()) m_dac.closeStream();- return;- }+ try {+ m_dac.openStream( &oParams, NULL, RTAUDIO_SINT16, frequency, &chunkSize, &cbMix, (void *)this );+ m_dac.startStream();+ }+ catch ( RtAudioError& e ) {+ fprintf(stderr,"%s\n", e.getMessage().c_str());+ if(m_dac.isStreamOpen()) m_dac.closeStream();+ return;+ } } DeviceAudioRt::~DeviceAudioRt() { if(m_dac.isStreamOpen()) m_dac.closeStream(); delete [] ma_sound;- for( map<unsigned int,AudioSample*>::iterator it=mm_sample.begin(); it!=mm_sample.end(); ++it)- delete it->second;- mm_sample.clear();+ for( map<uint64_t,AudioSample*>::iterator it=mm_sample.begin(); it!=mm_sample.end(); ++it)+ delete it->second;+ mm_sample.clear(); } -unsigned int DeviceAudioRt::sampleFromMemory(const AudioSample & sample, float volume) {- AudioSample * pSample = new AudioSample(sample);- if(volume!=1.0f) pSample->volume(volume);- pSample->bitsPerSample(16);- mm_sample.insert(make_pair(++m_sampleCounter,pSample));- return m_sampleCounter;+uint64_t DeviceAudioRt::sampleFromMemory(const AudioSample & sample, float volume) {+ AudioSample * pSample = new AudioSample(sample);+ if(volume!=1.0f) pSample->volume(volume);+ pSample->bitsPerSample(16);+ mm_sample.insert(make_pair(++m_uniqueCounter,pSample));+ return m_uniqueCounter; } -bool DeviceAudioRt::sampleDestroy(unsigned int sample) {+bool DeviceAudioRt::sampleDestroy(uint64_t sample) { // look for sample:- map<unsigned int,AudioSample*>::iterator iter=mm_sample.find(sample);+ map<uint64_t,AudioSample*>::iterator iter=mm_sample.find(sample); if( iter == mm_sample.end() ) return false;- // stop currently playing sounds referring to this sample:- for (unsigned int i=0; i<m_nSound; ++i ) if(ma_sound[i].sample == iter->second)- ma_sound[i].isPlaying=false;- // cleanup:- delete iter->second;- if(iter->first==m_sampleCounter) --m_sampleCounter;- mm_sample.erase(iter);- return true;+ // stop currently playing sounds referring to this sample:+ for (unsigned int i=0; i<m_nSound; ++i ) if(ma_sound[i].sample == iter->second)+ ma_sound[i].isPlaying=false;+ // cleanup:+ delete iter->second;+ mm_sample.erase(iter);+ return true; } -const AudioSample* DeviceAudioRt::sample(unsigned int handle) const { - map<unsigned int,AudioSample*>::const_iterator it=mm_sample.find(handle);+const AudioSample* DeviceAudioRt::sample(uint64_t handle) const {+ map<uint64_t,AudioSample*>::const_iterator it=mm_sample.find(handle); if( it == mm_sample.end() ) return 0;- return it->second;+ return it->second; } --unsigned int DeviceAudioRt::soundPlay(unsigned int sample, float volumeL, float volumeR, float disparity, float pitch ) {- // look for sample:- map<unsigned int,AudioSample*>::iterator iter=mm_sample.find(sample);- if( iter == mm_sample.end() ) return 0; // no sample found+uint64_t DeviceAudioRt::soundPlay(uint64_t sample, float volumeL, float volumeR, float disparity, float pitch) { // look for an empty (or finished) sound track unsigned int i; for ( i=0; i<m_nSound; ++i ) if (!ma_sound[i].isPlaying) break;- if ( i == m_nSound ) return 0; // no empty slot found+ return soundPlayOn(i,sample,volumeL,volumeR,disparity,pitch);+} - unsigned int sampleRate = iter->second->sampleRate();- if(sampleRate!=m_freqOut) pitch*=(float)sampleRate/(float)m_freqOut;- +uint64_t DeviceAudioRt::soundPlayOn(unsigned int i, uint64_t sample, float volumeL, float volumeR, float disparity, float pitch) {+ if ( i >= m_nSound ) return 0; // invalid, or no empty slot found++ // look for sample:+ map<uint64_t,AudioSample*>::iterator iter=mm_sample.find(sample);+ if( iter == mm_sample.end() ) return 0; // no sample found++ uint64_t uniqueHandle = UH_PACK_UNIQUE_HANDLE(++m_uniqueCounter, i);+ unsigned int sampleRate = iter->second->sampleRate();+ if(sampleRate!=m_freqOut) pitch*=(float)sampleRate/(float)m_freqOut;+ // put the sample data in the slot and play it ma_sound[i].sample = iter->second; ma_sound[i].dlen = iter->second->frames();@@ -127,110 +137,125 @@ ma_sound[i].pitch=fabs(pitch); ma_sound[i].isLoop=false; ma_sound[i].isPlaying=true;- return i+1;+ ma_sound[i].uniqueId=UH_UNPACK_UNIQUE_ID(uniqueHandle);+ return uniqueHandle; } -unsigned int DeviceAudioRt::soundLoop(unsigned int sample, float volumeL, float volumeR, float disparity, float pitch ) {- unsigned int ret=soundPlay(sample,volumeL,volumeR,disparity, pitch);- if(ret) ma_sound[ret-1].isLoop=true;- return ret;+uint64_t DeviceAudioRt::soundLoop(uint64_t sample, float volumeL, float volumeR, float disparity, float pitch ) {+ uint64_t uniqueHandle=soundPlay(sample,volumeL,volumeR,disparity, pitch);+ if(uniqueHandle) ma_sound[UH_UNPACK_PAYLOAD(uniqueHandle)].isLoop=true;+ return uniqueHandle; } -bool DeviceAudioRt::soundUpdate(unsigned int sound, float volumeL, float volumeR, float disparity, float pitch ) {- if(!sound || (sound>m_nSound) || !ma_sound[sound-1].isPlaying) return false;- ma_sound[--sound].volL=volumeL;+uint64_t DeviceAudioRt::soundLoopOn(unsigned int i, uint64_t sample, float volumeL, float volumeR, float disparity, float pitch) {+ uint64_t uniqueHandle=soundPlayOn(i,sample,volumeL,volumeR,disparity, pitch);+ if(uniqueHandle) ma_sound[UH_UNPACK_PAYLOAD(uniqueHandle)].isLoop=true;+ return uniqueHandle;+}++bool DeviceAudioRt::soundUpdate(uint64_t uniqueHandle, float volumeL, float volumeR, float disparity, float pitch ) {+ unsigned int sound = UH_UNPACK_PAYLOAD(uniqueHandle);+ if((sound>=m_nSound) || !ma_sound[sound].isPlaying || ma_sound[sound].uniqueId!=UH_UNPACK_UNIQUE_ID(uniqueHandle)) return false;+ ma_sound[sound].volL=volumeL; ma_sound[sound].volR=volumeR; ma_sound[sound].disparity=disparity;- unsigned int sampleRate = ma_sound[sound].sample->sampleRate();- if(sampleRate!=m_freqOut) pitch*=(float)sampleRate/(float)m_freqOut;+ unsigned int sampleRate = ma_sound[sound].sample->sampleRate();+ if(sampleRate!=m_freqOut) pitch*=(float)sampleRate/(float)m_freqOut; ma_sound[sound].pitch=fabs(pitch); return true; } -bool DeviceAudioRt::soundStop(unsigned int sound) {- if(!sound||(sound>m_nSound)||!ma_sound[sound-1].isPlaying) return false;- ma_sound[sound-1].isPlaying=false;+bool DeviceAudioRt::soundStop(uint64_t uniqueHandle) {+ unsigned int sound = UH_UNPACK_PAYLOAD(uniqueHandle);+ if((sound>=m_nSound) || !ma_sound[sound].isPlaying || ma_sound[sound].uniqueId!=UH_UNPACK_UNIQUE_ID(uniqueHandle)) return false;+ ma_sound[sound].isPlaying=false; return true; } void DeviceAudioRt::soundStop() {- for (unsigned int i=0; i<m_nSound; ++i )- ma_sound[i].isPlaying=false;+ for (unsigned int i=0; i<m_nSound; ++i )+ ma_sound[i].isPlaying=false; } unsigned int DeviceAudioRt::soundActive() const {- if(!const_cast<RtAudio*>(&m_dac)->isStreamRunning() ) return 0;+ if(!const_cast<RtAudio*>(&m_dac)->isStreamRunning() ) return 0; unsigned int ret = 0, i; for ( i=0; i<m_nSound; ++i ) if (ma_sound[i].isPlaying) ++ret; return ret; } +bool DeviceAudioRt::soundActive(uint64_t uniqueHandle) {+ unsigned int sound = UH_UNPACK_PAYLOAD(uniqueHandle);+ if((sound>=m_nSound) || ma_sound[sound].uniqueId!=UH_UNPACK_UNIQUE_ID(uniqueHandle)) return false;+ return ma_sound[sound].isPlaying;+}+ int DeviceAudioRt::mixOutputFloat(signed short *outputBuffer, unsigned int nFrames) { for(unsigned int j=0; j<nFrames; ++j) { float left=0.0f; float right=0.0f; for (unsigned int i=0; i<m_nSound; ++i ) if(ma_sound[i].isPlaying) {- unsigned int nChannels = ma_sound[i].sample->channels();+ unsigned int nChannels = ma_sound[i].sample->channels(); if((ma_sound[i].pitch==1.0f)&&!ma_sound[i].disparity) { // use optimized default mixing: unsigned int currPos=ma_sound[i].dpos+j; if(ma_sound[i].isLoop) currPos%=ma_sound[i].dlen; else if(currPos >= ma_sound[i].dlen) continue;- currPos*=ma_sound[i].sample->sizeFrame();- float dataL = (float)(*((signed short *)(&ma_sound[i].sample->data()[currPos])));+ currPos*=ma_sound[i].sample->sizeFrame();+ float dataL = (float)(*((signed short *)(&ma_sound[i].sample->data()[currPos]))); left += dataL * m_volL*ma_sound[i].volL; float dataR = (nChannels>1) ? (float)(*((signed short *)(&ma_sound[i].sample->data()[currPos+2]))) : dataL;- right+= dataR * m_volR*ma_sound[i].volR; + right+= dataR * m_volR*ma_sound[i].volR; } else { // use nearest sample and disparity: double fract=ma_sound[i].dpos+j*ma_sound[i].pitch; unsigned int currPos=(unsigned int)fract;- fract = fmod(fract,1.0);+ fract = fmod(fract,1.0); int currPosL= (ma_sound[i].disparity<0.0f) ? currPos+int(m_freqOut*ma_sound[i].disparity) : currPos; int currPosR= (ma_sound[i].disparity>0.0f) ? currPos-int(m_freqOut*ma_sound[i].disparity) : currPos;- if(nChannels>1) currPosR+=sizeof(signed short); // use second channel+ if(nChannels>1) currPosR+=sizeof(signed short); // use second channel if(ma_sound[i].isLoop) {- currPosL+=ma_sound[i].dlen;- currPosL%=ma_sound[i].dlen;- currPosR+=ma_sound[i].dlen;- currPosR%=ma_sound[i].dlen;- }- if(currPosL<0) {- // do nothing- }- else if((unsigned int)currPosL+1 < ma_sound[i].dlen) {- currPosL*=ma_sound[i].sample->sizeFrame();- float dataL = (1.0f-(float)fract)*(float)(*((signed short *)(&ma_sound[i].sample->data()[currPosL])))- + (float)fract*(float)(*((signed short *)(&ma_sound[i].sample->data()[currPosL+ma_sound[i].sample->sizeFrame()])));- left += dataL * m_volL*ma_sound[i].volL;- }- else if((unsigned int)currPosL+1 == ma_sound[i].dlen) {- currPosL*=ma_sound[i].sample->sizeFrame();- float dataL = (float)(*((signed short *)(&ma_sound[i].sample->data()[currPosL])));- left += dataL * m_volL*ma_sound[i].volL;- }- - if(currPosR<0) {- // do nothing- }- else if((unsigned int)currPosR+1 < ma_sound[i].dlen) {- currPosR*=ma_sound[i].sample->sizeFrame();- float dataR = (1.0f-(float)fract)*(float)(*((signed short *)(&ma_sound[i].sample->data()[currPosR])))- + (float)fract*(float)(*((signed short *)(&ma_sound[i].sample->data()[currPosR+ma_sound[i].sample->sizeFrame()])));- right += dataR * m_volR*ma_sound[i].volR;- }- else if((unsigned int)currPosR+1 == ma_sound[i].dlen) {- currPosR*=ma_sound[i].sample->sizeFrame();- float dataR = (float)(*((signed short *)(&ma_sound[i].sample->data()[currPosR])));- right += dataR * m_volR*ma_sound[i].volR;- }+ currPosL+=ma_sound[i].dlen;+ currPosL%=ma_sound[i].dlen;+ currPosR+=ma_sound[i].dlen;+ currPosR%=ma_sound[i].dlen;+ }+ if(currPosL<0) {+ // do nothing+ }+ else if((unsigned int)currPosL+1 < ma_sound[i].dlen) {+ currPosL*=ma_sound[i].sample->sizeFrame();+ float dataL = (1.0f-(float)fract)*(float)(*((signed short *)(&ma_sound[i].sample->data()[currPosL])))+ + (float)fract*(float)(*((signed short *)(&ma_sound[i].sample->data()[currPosL+ma_sound[i].sample->sizeFrame()])));+ left += dataL * m_volL*ma_sound[i].volL;+ }+ else if((unsigned int)currPosL+1 == ma_sound[i].dlen) {+ currPosL*=ma_sound[i].sample->sizeFrame();+ float dataL = (float)(*((signed short *)(&ma_sound[i].sample->data()[currPosL])));+ left += dataL * m_volL*ma_sound[i].volL;+ }++ if(currPosR<0) {+ // do nothing+ }+ else if((unsigned int)currPosR+1 < ma_sound[i].dlen) {+ currPosR*=ma_sound[i].sample->sizeFrame();+ float dataR = (1.0f-(float)fract)*(float)(*((signed short *)(&ma_sound[i].sample->data()[currPosR])))+ + (float)fract*(float)(*((signed short *)(&ma_sound[i].sample->data()[currPosR+ma_sound[i].sample->sizeFrame()])));+ right += dataR * m_volR*ma_sound[i].volR;+ }+ else if((unsigned int)currPosR+1 == ma_sound[i].dlen) {+ currPosR*=ma_sound[i].sample->sizeFrame();+ float dataR = (float)(*((signed short *)(&ma_sound[i].sample->data()[currPosR])));+ right += dataR * m_volR*ma_sound[i].volR;+ } } } // clamp and set output: outputBuffer[2*j] = left>SHRT_MAX ? SHRT_MAX : left<SHRT_MIN ? SHRT_MIN : (signed short)left; outputBuffer[2*j+1] = right>SHRT_MAX ? SHRT_MAX : right<SHRT_MIN ? SHRT_MIN : (signed short)right; }- // calculate new pos:+ // calculate new pos: for (unsigned int i=0; i<m_nSound; ++i ) { if(ma_sound[i].pitch==1.0f) ma_sound[i].dpos += nFrames; else ma_sound[i].dpos += (unsigned int)(nFrames*ma_sound[i].pitch);@@ -239,5 +264,5 @@ else if(ma_sound[i].dpos>ma_sound[i].dlen+2*abs(int(m_freqOut*-ma_sound[i].disparity))) ma_sound[i].isPlaying=false; }- return 0;+ return 0; }
cbits/proAudioRt.h view
@@ -1,12 +1,12 @@ #include "proAudio.h"-#include <RtAudio.h>+#include "RtAudio.h" #include <map> /** @file proAudioRt.h \brief RtAudio backend of proteaAudio \author Gerald Franz, www.viremo.de \version 0.6-*/ +*/ struct _AudioTrack; @@ -15,31 +15,31 @@ independent volume control for both channels, and user-defined time shifts between the channels. */ class DeviceAudioRt : public DeviceAudio { public:- ///creates audio device+ ///creates audio device /** Use this method instead of a constructor. \param nTracks (optional) the maximum number of sounds that are played parallely. Computation time is linearly correlated to this factor. \param frequency (optional) sample frequency of the playback in Hz. 22050 corresponds to FM radio 44100 is CD quality. Computation time is linearly correlated to this factor. \param chunkSize (optional) the number of bytes that are sent to the sound card at once. Low numbers lead to smaller latencies but need more computation time (thread switches). If a too small number is chosen, the sounds might not be played continuously. The default value 512 guarantees a good latency below 40 ms at 22050 Hz sample frequency. \return a pointer to an audio device object in case of success- Note that the parameters are only handled when calling for the first time. Afterwards always the same object is returned until an explicit destroy() is called.+ Note that the parameters are only handled when calling for the first time. Afterwards always the same object is returned until an explicit destroy() is called. */ static DeviceAudio* create(unsigned int nTracks=8, unsigned int frequency=22050, unsigned int chunkSize=1024); /// converts a sound sample to internal audio format, returns handle- virtual unsigned int sampleFromMemory(const AudioSample & sample, float volume=1.0f);- /// deletes a previously created sound sample resource identified by its handle- virtual bool sampleDestroy(unsigned int sample);- /// allows read access to a sample identified by its handle- virtual const AudioSample* sample(unsigned int handle) const;+ virtual uint64_t sampleFromMemory(const AudioSample & sample, float volume=1.0f);+ /// deletes a previously created sound sample resource identified by its handle+ virtual bool sampleDestroy(uint64_t sample);+ /// allows read access to a sample identified by its handle+ virtual const AudioSample* sample(uint64_t handle) const; - /// plays a specified sample once and sets its parameters+ /// plays a specified sample once and sets its parameters /** \param sample a sample handle returned by a previous load() call \param volumeL (optional) left volume \param volumeR (optional) right volume \param disparity (optional) time difference between left and right channel in seconds. Use negative values to specify a delay for the left channel, positive for the right. \param pitch (optional) pitch factor for playback. 0.5 corresponds to one octave below, 2.0 to one above the original sample. \return a handle to the currently played sound or 0 in case of error */- virtual unsigned int soundPlay(unsigned int sample, float volumeL=1.0f, float volumeR=1.0f, float disparity=0.0f, float pitch=1.0f );+ virtual uint64_t soundPlay(uint64_t sample, float volumeL=1.0f, float volumeR=1.0f, float disparity=0.0f, float pitch=1.0f ); /** plays a specified sample continuously and sets its parameters \param sample a sample handle returned by a previous load() call \param volumeL (optional) left volume@@ -47,7 +47,25 @@ \param disparity (optional) time difference between left and right channel in seconds. Use negative values to specify a delay for the left channel, positive for the right. \param pitch (optional) pitch factor for playback. 0.5 corresponds to one octave below, 2.0 to one above the original sample. \return a handle to the currently played sound or 0 in case of error */- virtual unsigned int soundLoop(unsigned int sample, float volumeL=1.0f, float volumeR=1.0f, float disparity=0.0f, float pitch=1.0f );+ virtual uint64_t soundLoop(uint64_t sample, float volumeL=1.0f, float volumeR=1.0f, float disparity=0.0f, float pitch=1.0f );+ /// plays a specified sample once on a specific channel and sets its parameters+ /** \param i a channel number with the first channel starting at 0 + \param sample a sample handle returned by a previous load() call+ \param volumeL (optional) left volume+ \param volumeR (optional) right volume+ \param disparity (optional) time difference between left and right channel in seconds. Use negative values to specify a delay for the left channel, positive for the right.+ \param pitch (optional) pitch factor for playback. 0.5 corresponds to one octave below, 2.0 to one above the original sample.+ \return a handle to the currently played sound or 0 in case of error */+ virtual uint64_t soundPlayOn(unsigned int i, uint64_t sample, float volumeL=1.0f, float volumeR=1.0f, float disparity=0.0f, float pitch=1.0f );+ /// plays a specified sample continuously on a specific channel and sets its parameters+ /** \param i a channel number with the first channel starting at 0 + \param sample a sample handle returned by a previous load() call+ \param volumeL (optional) left volume+ \param volumeR (optional) right volume+ \param disparity (optional) time difference between left and right channel in seconds. Use negative values to specify a delay for the left channel, positive for the right.+ \param pitch (optional) pitch factor for playback. 0.5 corresponds to one octave below, 2.0 to one above the original sample.+ \return a handle to the currently played sound or 0 in case of error */+ virtual uint64_t soundLoopOn(unsigned int i, uint64_t sample, float volumeL=1.0f, float volumeR=1.0f, float disparity=0.0f, float pitch=1.0f ); /// updates parameters of a specified sound /** \param sound handle of a currently active sound \param volumeL left volume@@ -55,34 +73,36 @@ \param disparity (optional) time difference between left and right channel in seconds. Use negative values to specify a delay for the left channel, positive for the right. \param pitch (optional) pitch factor for playback. 0.5 corresponds to one octave below, 2.0 to one above the original sample. \return true in case the parameters have been updated successfully */- virtual bool soundUpdate(unsigned int sound, float volumeL, float volumeR, float disparity=0.0f, float pitch=1.0f );+ virtual bool soundUpdate(uint64_t sound, float volumeL, float volumeR, float disparity=0.0f, float pitch=1.0f ); /// stops a specified sound immediately- virtual bool soundStop(unsigned int sound);+ virtual bool soundStop(uint64_t sound); /// stops all sounds immediately virtual void soundStop();- /// returns number of currently active sounds- virtual unsigned soundActive() const;+ /// returns number of currently active sounds+ virtual unsigned soundActive() const;+ /// check if a specified sound is active+ virtual bool soundActive(uint64_t sound); protected: /// constructor. Use the create() method instead DeviceAudioRt(unsigned int nTracks, unsigned int frequency, unsigned int chunkSize); /// destructor. Use the destroy() method instead virtual ~DeviceAudioRt();- /// mixes tracks to a single output stream- int mixOutputFloat(signed short *outputBuffer, unsigned int nFrames);+ /// mixes tracks to a single output stream+ int mixOutputFloat(signed short *outputBuffer, unsigned int nFrames); - /// stores loaded sound samples- std::map<unsigned int, AudioSample*> mm_sample;- /// stores maximum sample id- unsigned int m_sampleCounter;+ /// stores loaded sound samples+ std::map<uint64_t, AudioSample*> mm_sample;+ /// counter for unique sample and sound id's+ uint64_t m_uniqueCounter; /// stores sounds to be mixed _AudioTrack * ma_sound; /// stores number of parallel sounds unsigned int m_nSound;- /// audio manager- RtAudio m_dac;+ /// audio manager+ RtAudio m_dac; /// mixer callback static int cbMix(void *outputBuffer, void *inputBuffer, unsigned int nFrames, double streamTime, RtAudioStreamStatus status, void *data) {- return static_cast<DeviceAudioRt*>(data)->mixOutputFloat((signed short*)outputBuffer, nFrames); }+ return static_cast<DeviceAudioRt*>(data)->mixOutputFloat((signed short*)outputBuffer, nFrames); } };
cbits/proteaaudio_binding.cpp view
@@ -1,8 +1,25 @@-#include "proteaaudio_binding.h"-#include "proAudioRt.h" #include <cstring> #include <cstdlib>+#include "proteaaudio_binding.h" +// check if either PROTEAAUDIO_SDL or PROTEAAUDIO_RT is set+#if defined PROTEAAUDIO_SDL && defined PROTEAAUDIO_RT+#error "both PROTEAAUDIO_SDL and PROTEAAUDIO_RT is defined"+#endif++#if !defined PROTEAAUDIO_SDL && !defined PROTEAAUDIO_RT+#error "neither PROTEAAUDIO_SDL and PROTEAAUDIO_RT is defined"+#endif++// import backend api+#ifdef PROTEAAUDIO_RT+#include "proAudioRt.h"+#endif++#ifdef PROTEAAUDIO_SDL+#include "proAudioSdl.h"+#endif+ using namespace std; typedef struct {@@ -56,7 +73,14 @@ // generic int initAudio(int nTracks, int frequency, int chunkSize) {+ if (nTracks < 1 || nTracks > (1<<UH_PAYLOAD_BIT_WIDTH) )+ return 0;+#ifdef PROTEAAUDIO_SDL+ DeviceAudio* pAudio = DeviceAudioSdl::create(nTracks, frequency, chunkSize);+#endif+#ifdef PROTEAAUDIO_RT DeviceAudio* pAudio = DeviceAudioRt::create(nTracks, frequency, chunkSize);+#endif return pAudio != 0; } @@ -79,7 +103,7 @@ sample_t sampleFromFile(char* filename, float volume) { DeviceAudio & audio = DeviceAudio::singleton(); if((&audio) == 0) return 0;- return (int)audio.sampleFromFile(filename, volume);+ return audio.sampleFromFile(filename, volume); } sample_t _sampleFromMemoryPcm(char *pcm_data, int size, int channels, int sampleRate, int bitsPerSample, float volume) {@@ -90,18 +114,18 @@ memcpy(data, pcm_data, size); AudioSample * pSample = new AudioSample(data, size, channels, sampleRate, bitsPerSample); if(!pSample) return 0;- unsigned int ret = audio.sampleFromMemory(*pSample, volume);+ sample_t ret = audio.sampleFromMemory(*pSample, volume); delete pSample;- return (int)ret;+ return ret; } sample_t _sampleFromMemoryWav(char *data, int size, float volume) { DeviceAudio & audio = DeviceAudio::singleton(); if((&audio) == 0) return 0; AudioSample* pSample = loadWavFromMemory((unsigned char*)data, size); if(!pSample) return 0;- unsigned int ret = audio.sampleFromMemory(*pSample, volume);+ sample_t ret = audio.sampleFromMemory(*pSample, volume); delete pSample;- return (int)ret;+ return ret; } sample_t _sampleFromMemoryOgg(char *data, int size, float volume) {@@ -109,12 +133,12 @@ if((&audio) == 0) return 0; AudioSample* pSample = loadOggFromMemory((unsigned char*)data, size); if(!pSample) return 0;- unsigned int ret = audio.sampleFromMemory(*pSample, volume);+ sample_t ret = audio.sampleFromMemory(*pSample, volume); delete pSample;- return (int)ret;+ return ret; } -int soundActive() {+int soundActiveAll() { DeviceAudio & audio = DeviceAudio::singleton(); if((&audio) == 0) return 0; return (int)audio.soundActive();@@ -126,6 +150,12 @@ audio.soundStop(); } +int sampleDestroy(uint64_t sample) {+ DeviceAudio & audio = DeviceAudio::singleton();+ if((&audio) == 0) return false;+ return audio.sampleDestroy(sample);+}+ // sound sound_t soundLoop(sample_t sample, float volumeL, float volumeR, float disparity, float pitch) { DeviceAudio & audio = DeviceAudio::singleton();@@ -133,12 +163,24 @@ return audio.soundLoop(sample, volumeL, volumeR, disparity, pitch); } -sound_t soundPlay(sample_t sample, float volumeL, float volumeR, float disparity, float pitch) {+sound_t soundPlay(sample_t sample, float volumeL, float volumeR, float disparity, float pitch) { DeviceAudio & audio = DeviceAudio::singleton(); if((&audio) == 0) return 0; return audio.soundPlay(sample, volumeL, volumeR, disparity, pitch); } +sound_t soundLoopOn(unsigned int track, sample_t sample, float volumeL, float volumeR, float disparity, float pitch) {+ DeviceAudio & audio = DeviceAudio::singleton();+ if((&audio) == 0) return 0;+ return audio.soundLoopOn(track, sample, volumeL, volumeR, disparity, pitch); +}++sound_t soundPlayOn(unsigned int track, sample_t sample, float volumeL, float volumeR, float disparity, float pitch) {+ DeviceAudio & audio = DeviceAudio::singleton();+ if((&audio) == 0) return 0;+ return audio.soundPlayOn(track, sample, volumeL, volumeR, disparity, pitch); +}+ int soundUpdate(sound_t sound, float volumeL, float volumeR, float disparity, float pitch) { DeviceAudio & audio = DeviceAudio::singleton(); if((&audio) == 0) return 0;@@ -149,4 +191,10 @@ DeviceAudio & audio = DeviceAudio::singleton(); if((&audio) == 0) return 0; return audio.soundStop(sound);+}++int soundActive(sound_t sound) {+ DeviceAudio & audio = DeviceAudio::singleton();+ if((&audio) == 0) return 0;+ return audio.soundActive(sound); }
cbits/proteaaudio_binding.h view
@@ -2,9 +2,11 @@ extern "C" { #endif -typedef int sample_t;-typedef int sound_t;+#include <stdint.h> +typedef uint64_t sample_t;+typedef uint64_t sound_t;+ int initAudio(int nTracks, int frequency, int chunkSize); void finishAudio(); int loaderAvailable(char* suffix);@@ -13,12 +15,16 @@ sample_t _sampleFromMemoryWav(char *data, int size, float volume); sample_t _sampleFromMemoryOgg(char *data, int size, float volume); sample_t sampleFromFile(char* filename, float volume);-int soundActive();+int sampleDestroy(uint64_t sample);+int soundActiveAll(); void soundStopAll(); sound_t soundLoop(sample_t sample, float volumeL, float volumeR, float disparity, float pitch); sound_t soundPlay(sample_t sample, float volumeL, float volumeR, float disparity, float pitch);+sound_t soundPlayOn(unsigned int track, sample_t sample, float volumeL, float volumeR, float disparity, float pitch);+sound_t soundLoopOn(unsigned int track, sample_t sample, float volumeL, float volumeR, float disparity, float pitch); int soundUpdate(sound_t sound, float volumeL, float volumeR, float disparity, float pitch); int soundStop(sound_t sound);+int soundActive(sound_t sound); #ifdef __cplusplus } #endif
cbits/stb_vorbis.c view
@@ -1,4 +1,4 @@-// Ogg Vorbis audio decoder - v1.14 - public domain+// Ogg Vorbis audio decoder - v1.19 - public domain // http://nothings.org/stb_vorbis/ // // Original version written by Sean Barrett in 2007.@@ -26,17 +26,23 @@ // Terje Mathisen Niklas Frykholm Andy Hill // Casey Muratori John Bolton Gargaj // Laurent Gomila Marc LeBlanc Ronny Chevalier-// Bernhard Wodo Evan Balster alxprd@github+// Bernhard Wodo Evan Balster github:alxprd // Tom Beaumont Ingo Leitgeb Nicolas Guillemot // Phillip Bennefall Rohit Thiago Goulart-// manxorist@github saga musix github:infatum-// Timur Gagiev+// github:manxorist saga musix github:infatum+// Timur Gagiev Maxwell Koo Peter Waller+// github:audinowho Dougall Johnson // // Partial history:+// 1.19 - 2020-02-05 - warnings+// 1.18 - 2020-02-02 - fix seek bugs; parse header comments; misc warnings etc.+// 1.17 - 2019-07-08 - fix CVE-2019-13217..CVE-2019-13223 (by ForAllSecure)+// 1.16 - 2019-03-04 - fix warnings+// 1.15 - 2019-02-07 - explicit failure if Ogg Skeleton data is found // 1.14 - 2018-02-11 - delete bogus dealloca usage // 1.13 - 2018-01-29 - fix truncation of last frame (hopefully) // 1.12 - 2017-11-21 - limit residue begin/end to blocksize/2 to avoid large temp allocs in bad/corrupt files-// 1.11 - 2017-07-23 - fix MinGW compilation +// 1.11 - 2017-07-23 - fix MinGW compilation // 1.10 - 2017-03-03 - more robust seeking; fix negative ilog(); clear error in open_memory // 1.09 - 2016-04-04 - back out 'truncation of last frame' fix from previous version // 1.08 - 2016-04-02 - warnings; setup memory leaks; truncation of last frame@@ -127,9 +133,20 @@ int max_frame_size; } stb_vorbis_info; +typedef struct+{+ char *vendor;++ int comment_list_length;+ char **comment_list;+} stb_vorbis_comment;+ // get general information about the file extern stb_vorbis_info stb_vorbis_get_info(stb_vorbis *f); +// get ogg comments+extern stb_vorbis_comment stb_vorbis_get_comment(stb_vorbis *f);+ // get the last error detected (clears it, too) extern int stb_vorbis_get_error(stb_vorbis *f); @@ -253,7 +270,7 @@ // create an ogg vorbis decoder from an open FILE *, looking for a stream at // the _current_ seek point (ftell). on failure, returns NULL and sets *error. // note that stb_vorbis must "own" this stream; if you seek it in between-// calls to stb_vorbis, it will become confused. Morever, if you attempt to+// calls to stb_vorbis, it will become confused. Moreover, if you attempt to // perform stb_vorbis_seek_*() operations on this file, it will assume it // owns the _entire_ rest of the file after the start point. Use the next // function, stb_vorbis_open_file_section(), to limit it.@@ -374,7 +391,8 @@ VORBIS_invalid_first_page, VORBIS_bad_packet_type, VORBIS_cant_find_last_page,- VORBIS_seek_failed+ VORBIS_seek_failed,+ VORBIS_ogg_skeleton_not_supported }; @@ -755,6 +773,10 @@ unsigned int temp_memory_required; unsigned int setup_temp_memory_required; + char *vendor;+ int comment_list_length;+ char **comment_list;+ // input config #ifndef STB_VORBIS_NO_STDIO FILE *f;@@ -770,8 +792,11 @@ uint8 push_mode; + // the page to seek to when seeking to start, may be zero uint32 first_audio_page_offset; + // p_first is the page on which the first audio packet ends+ // (but not necessarily the page on which it starts) ProbedPage p_first, p_last; // memory management@@ -820,7 +845,7 @@ int current_loc_valid; // per-blocksize precomputed data- + // twiddle factors float *A[2],*B[2],*C[2]; float *window[2];@@ -884,7 +909,7 @@ #define array_size_required(count,size) (count*(sizeof(void *)+(size))) #define temp_alloc(f,size) (f->alloc.alloc_buffer ? setup_temp_malloc(f,size) : alloca(size))-#define temp_free(f,p) 0+#define temp_free(f,p) (void)0 #define temp_alloc_save(f) ((f)->temp_offset) #define temp_alloc_restore(f,p) ((f)->temp_offset = (p)) @@ -905,7 +930,7 @@ static void *setup_malloc(vorb *f, int sz) {- sz = (sz+3) & ~3;+ sz = (sz+7) & ~7; // round up to nearest 8 for alignment of future allocs. f->setup_memory_required += sz; if (f->alloc.alloc_buffer) { void *p = (char *) f->alloc.alloc_buffer + f->setup_offset;@@ -924,7 +949,7 @@ static void *setup_temp_malloc(vorb *f, int sz) {- sz = (sz+3) & ~3;+ sz = (sz+7) & ~7; // round up to nearest 8 for alignment of future allocs. if (f->alloc.alloc_buffer) { if (f->temp_offset - sz < f->setup_offset) return NULL; f->temp_offset -= sz;@@ -1073,7 +1098,7 @@ assert(z >= 0 && z < 32); available[z] = 0; add_entry(c, bit_reverse(res), i, m++, len[i], values);- // propogate availability up the tree+ // propagate availability up the tree if (z != len[i]) { assert(len[i] >= 0 && len[i] < 32); for (y=len[i]; y > z; --y) {@@ -1142,7 +1167,7 @@ if (!c->sparse) { int k = 0; for (i=0; i < c->entries; ++i)- if (include_in_sort(c, lengths[i])) + if (include_in_sort(c, lengths[i])) c->sorted_codewords[k++] = bit_reverse(c->codewords[i]); assert(k == c->sorted_entries); } else {@@ -1199,8 +1224,10 @@ int r = (int) floor(exp((float) log((float) entries) / dim)); if ((int) floor(pow((float) r+1, dim)) <= entries) // (int) cast for MinGW warning; ++r; // floor() to avoid _ftol() when non-CRT- assert(pow((float) r+1, dim) > entries);- assert((int) floor(pow((float) r, dim)) <= entries); // (int),floor() as above+ if (pow((float) r+1, dim) <= entries)+ return -1;+ if ((int) floor(pow((float) r, dim)) > entries)+ return -1; return r; } @@ -1323,7 +1350,7 @@ return 1; } - #ifndef STB_VORBIS_NO_STDIO + #ifndef STB_VORBIS_NO_STDIO if (fread(data, n, 1, z->f) == 1) return 1; else {@@ -1398,12 +1425,15 @@ static int start_page_no_capturepattern(vorb *f) { uint32 loc0,loc1,n;+ if (f->first_decode && !IS_PUSH_MODE(f)) {+ f->p_first.page_start = stb_vorbis_get_file_offset(f) - 4;+ } // stream structure version if (0 != get8(f)) return error(f, VORBIS_invalid_stream_structure_version); // header flag f->page_flag = get8(f); // absolute granule position- loc0 = get32(f); + loc0 = get32(f); loc1 = get32(f); // @TODO: validate loc0,loc1 as valid positions? // stream serial number -- vorbis doesn't interleave, so discard@@ -1434,15 +1464,12 @@ } if (f->first_decode) { int i,len;- ProbedPage p; len = 0; for (i=0; i < f->segment_count; ++i) len += f->segments[i]; len += 27 + f->segment_count;- p.page_start = f->first_audio_page_offset;- p.page_end = p.page_start + len;- p.last_decoded_sample = loc0;- f->p_first = p;+ f->p_first.page_end = f->p_first.page_start + len;+ f->p_first.last_decoded_sample = loc0; } f->next_seg = 0; return TRUE;@@ -1533,6 +1560,16 @@ return x; } +static int get32_packet(vorb *f)+{+ uint32 x;+ x = get8_packet(f);+ x += get8_packet(f) << 8;+ x += get8_packet(f) << 16;+ x += (uint32) get8_packet(f) << 24;+ return x;+}+ static void flush_packet(vorb *f) { while (get8_packet_raw(f) != EOP);@@ -1888,69 +1925,69 @@ // the following table is block-copied from the specification static float inverse_db_table[256] = {- 1.0649863e-07f, 1.1341951e-07f, 1.2079015e-07f, 1.2863978e-07f, - 1.3699951e-07f, 1.4590251e-07f, 1.5538408e-07f, 1.6548181e-07f, - 1.7623575e-07f, 1.8768855e-07f, 1.9988561e-07f, 2.1287530e-07f, - 2.2670913e-07f, 2.4144197e-07f, 2.5713223e-07f, 2.7384213e-07f, - 2.9163793e-07f, 3.1059021e-07f, 3.3077411e-07f, 3.5226968e-07f, - 3.7516214e-07f, 3.9954229e-07f, 4.2550680e-07f, 4.5315863e-07f, - 4.8260743e-07f, 5.1396998e-07f, 5.4737065e-07f, 5.8294187e-07f, - 6.2082472e-07f, 6.6116941e-07f, 7.0413592e-07f, 7.4989464e-07f, - 7.9862701e-07f, 8.5052630e-07f, 9.0579828e-07f, 9.6466216e-07f, - 1.0273513e-06f, 1.0941144e-06f, 1.1652161e-06f, 1.2409384e-06f, - 1.3215816e-06f, 1.4074654e-06f, 1.4989305e-06f, 1.5963394e-06f, - 1.7000785e-06f, 1.8105592e-06f, 1.9282195e-06f, 2.0535261e-06f, - 2.1869758e-06f, 2.3290978e-06f, 2.4804557e-06f, 2.6416497e-06f, - 2.8133190e-06f, 2.9961443e-06f, 3.1908506e-06f, 3.3982101e-06f, - 3.6190449e-06f, 3.8542308e-06f, 4.1047004e-06f, 4.3714470e-06f, - 4.6555282e-06f, 4.9580707e-06f, 5.2802740e-06f, 5.6234160e-06f, - 5.9888572e-06f, 6.3780469e-06f, 6.7925283e-06f, 7.2339451e-06f, - 7.7040476e-06f, 8.2047000e-06f, 8.7378876e-06f, 9.3057248e-06f, - 9.9104632e-06f, 1.0554501e-05f, 1.1240392e-05f, 1.1970856e-05f, - 1.2748789e-05f, 1.3577278e-05f, 1.4459606e-05f, 1.5399272e-05f, - 1.6400004e-05f, 1.7465768e-05f, 1.8600792e-05f, 1.9809576e-05f, - 2.1096914e-05f, 2.2467911e-05f, 2.3928002e-05f, 2.5482978e-05f, - 2.7139006e-05f, 2.8902651e-05f, 3.0780908e-05f, 3.2781225e-05f, - 3.4911534e-05f, 3.7180282e-05f, 3.9596466e-05f, 4.2169667e-05f, - 4.4910090e-05f, 4.7828601e-05f, 5.0936773e-05f, 5.4246931e-05f, - 5.7772202e-05f, 6.1526565e-05f, 6.5524908e-05f, 6.9783085e-05f, - 7.4317983e-05f, 7.9147585e-05f, 8.4291040e-05f, 8.9768747e-05f, - 9.5602426e-05f, 0.00010181521f, 0.00010843174f, 0.00011547824f, - 0.00012298267f, 0.00013097477f, 0.00013948625f, 0.00014855085f, - 0.00015820453f, 0.00016848555f, 0.00017943469f, 0.00019109536f, - 0.00020351382f, 0.00021673929f, 0.00023082423f, 0.00024582449f, - 0.00026179955f, 0.00027881276f, 0.00029693158f, 0.00031622787f, - 0.00033677814f, 0.00035866388f, 0.00038197188f, 0.00040679456f, - 0.00043323036f, 0.00046138411f, 0.00049136745f, 0.00052329927f, - 0.00055730621f, 0.00059352311f, 0.00063209358f, 0.00067317058f, - 0.00071691700f, 0.00076350630f, 0.00081312324f, 0.00086596457f, - 0.00092223983f, 0.00098217216f, 0.0010459992f, 0.0011139742f, - 0.0011863665f, 0.0012634633f, 0.0013455702f, 0.0014330129f, - 0.0015261382f, 0.0016253153f, 0.0017309374f, 0.0018434235f, - 0.0019632195f, 0.0020908006f, 0.0022266726f, 0.0023713743f, - 0.0025254795f, 0.0026895994f, 0.0028643847f, 0.0030505286f, - 0.0032487691f, 0.0034598925f, 0.0036847358f, 0.0039241906f, - 0.0041792066f, 0.0044507950f, 0.0047400328f, 0.0050480668f, - 0.0053761186f, 0.0057254891f, 0.0060975636f, 0.0064938176f, - 0.0069158225f, 0.0073652516f, 0.0078438871f, 0.0083536271f, - 0.0088964928f, 0.009474637f, 0.010090352f, 0.010746080f, - 0.011444421f, 0.012188144f, 0.012980198f, 0.013823725f, - 0.014722068f, 0.015678791f, 0.016697687f, 0.017782797f, - 0.018938423f, 0.020169149f, 0.021479854f, 0.022875735f, - 0.024362330f, 0.025945531f, 0.027631618f, 0.029427276f, - 0.031339626f, 0.033376252f, 0.035545228f, 0.037855157f, - 0.040315199f, 0.042935108f, 0.045725273f, 0.048696758f, - 0.051861348f, 0.055231591f, 0.058820850f, 0.062643361f, - 0.066714279f, 0.071049749f, 0.075666962f, 0.080584227f, - 0.085821044f, 0.091398179f, 0.097337747f, 0.10366330f, - 0.11039993f, 0.11757434f, 0.12521498f, 0.13335215f, - 0.14201813f, 0.15124727f, 0.16107617f, 0.17154380f, - 0.18269168f, 0.19456402f, 0.20720788f, 0.22067342f, - 0.23501402f, 0.25028656f, 0.26655159f, 0.28387361f, - 0.30232132f, 0.32196786f, 0.34289114f, 0.36517414f, - 0.38890521f, 0.41417847f, 0.44109412f, 0.46975890f, - 0.50028648f, 0.53279791f, 0.56742212f, 0.60429640f, - 0.64356699f, 0.68538959f, 0.72993007f, 0.77736504f, + 1.0649863e-07f, 1.1341951e-07f, 1.2079015e-07f, 1.2863978e-07f,+ 1.3699951e-07f, 1.4590251e-07f, 1.5538408e-07f, 1.6548181e-07f,+ 1.7623575e-07f, 1.8768855e-07f, 1.9988561e-07f, 2.1287530e-07f,+ 2.2670913e-07f, 2.4144197e-07f, 2.5713223e-07f, 2.7384213e-07f,+ 2.9163793e-07f, 3.1059021e-07f, 3.3077411e-07f, 3.5226968e-07f,+ 3.7516214e-07f, 3.9954229e-07f, 4.2550680e-07f, 4.5315863e-07f,+ 4.8260743e-07f, 5.1396998e-07f, 5.4737065e-07f, 5.8294187e-07f,+ 6.2082472e-07f, 6.6116941e-07f, 7.0413592e-07f, 7.4989464e-07f,+ 7.9862701e-07f, 8.5052630e-07f, 9.0579828e-07f, 9.6466216e-07f,+ 1.0273513e-06f, 1.0941144e-06f, 1.1652161e-06f, 1.2409384e-06f,+ 1.3215816e-06f, 1.4074654e-06f, 1.4989305e-06f, 1.5963394e-06f,+ 1.7000785e-06f, 1.8105592e-06f, 1.9282195e-06f, 2.0535261e-06f,+ 2.1869758e-06f, 2.3290978e-06f, 2.4804557e-06f, 2.6416497e-06f,+ 2.8133190e-06f, 2.9961443e-06f, 3.1908506e-06f, 3.3982101e-06f,+ 3.6190449e-06f, 3.8542308e-06f, 4.1047004e-06f, 4.3714470e-06f,+ 4.6555282e-06f, 4.9580707e-06f, 5.2802740e-06f, 5.6234160e-06f,+ 5.9888572e-06f, 6.3780469e-06f, 6.7925283e-06f, 7.2339451e-06f,+ 7.7040476e-06f, 8.2047000e-06f, 8.7378876e-06f, 9.3057248e-06f,+ 9.9104632e-06f, 1.0554501e-05f, 1.1240392e-05f, 1.1970856e-05f,+ 1.2748789e-05f, 1.3577278e-05f, 1.4459606e-05f, 1.5399272e-05f,+ 1.6400004e-05f, 1.7465768e-05f, 1.8600792e-05f, 1.9809576e-05f,+ 2.1096914e-05f, 2.2467911e-05f, 2.3928002e-05f, 2.5482978e-05f,+ 2.7139006e-05f, 2.8902651e-05f, 3.0780908e-05f, 3.2781225e-05f,+ 3.4911534e-05f, 3.7180282e-05f, 3.9596466e-05f, 4.2169667e-05f,+ 4.4910090e-05f, 4.7828601e-05f, 5.0936773e-05f, 5.4246931e-05f,+ 5.7772202e-05f, 6.1526565e-05f, 6.5524908e-05f, 6.9783085e-05f,+ 7.4317983e-05f, 7.9147585e-05f, 8.4291040e-05f, 8.9768747e-05f,+ 9.5602426e-05f, 0.00010181521f, 0.00010843174f, 0.00011547824f,+ 0.00012298267f, 0.00013097477f, 0.00013948625f, 0.00014855085f,+ 0.00015820453f, 0.00016848555f, 0.00017943469f, 0.00019109536f,+ 0.00020351382f, 0.00021673929f, 0.00023082423f, 0.00024582449f,+ 0.00026179955f, 0.00027881276f, 0.00029693158f, 0.00031622787f,+ 0.00033677814f, 0.00035866388f, 0.00038197188f, 0.00040679456f,+ 0.00043323036f, 0.00046138411f, 0.00049136745f, 0.00052329927f,+ 0.00055730621f, 0.00059352311f, 0.00063209358f, 0.00067317058f,+ 0.00071691700f, 0.00076350630f, 0.00081312324f, 0.00086596457f,+ 0.00092223983f, 0.00098217216f, 0.0010459992f, 0.0011139742f,+ 0.0011863665f, 0.0012634633f, 0.0013455702f, 0.0014330129f,+ 0.0015261382f, 0.0016253153f, 0.0017309374f, 0.0018434235f,+ 0.0019632195f, 0.0020908006f, 0.0022266726f, 0.0023713743f,+ 0.0025254795f, 0.0026895994f, 0.0028643847f, 0.0030505286f,+ 0.0032487691f, 0.0034598925f, 0.0036847358f, 0.0039241906f,+ 0.0041792066f, 0.0044507950f, 0.0047400328f, 0.0050480668f,+ 0.0053761186f, 0.0057254891f, 0.0060975636f, 0.0064938176f,+ 0.0069158225f, 0.0073652516f, 0.0078438871f, 0.0083536271f,+ 0.0088964928f, 0.009474637f, 0.010090352f, 0.010746080f,+ 0.011444421f, 0.012188144f, 0.012980198f, 0.013823725f,+ 0.014722068f, 0.015678791f, 0.016697687f, 0.017782797f,+ 0.018938423f, 0.020169149f, 0.021479854f, 0.022875735f,+ 0.024362330f, 0.025945531f, 0.027631618f, 0.029427276f,+ 0.031339626f, 0.033376252f, 0.035545228f, 0.037855157f,+ 0.040315199f, 0.042935108f, 0.045725273f, 0.048696758f,+ 0.051861348f, 0.055231591f, 0.058820850f, 0.062643361f,+ 0.066714279f, 0.071049749f, 0.075666962f, 0.080584227f,+ 0.085821044f, 0.091398179f, 0.097337747f, 0.10366330f,+ 0.11039993f, 0.11757434f, 0.12521498f, 0.13335215f,+ 0.14201813f, 0.15124727f, 0.16107617f, 0.17154380f,+ 0.18269168f, 0.19456402f, 0.20720788f, 0.22067342f,+ 0.23501402f, 0.25028656f, 0.26655159f, 0.28387361f,+ 0.30232132f, 0.32196786f, 0.34289114f, 0.36517414f,+ 0.38890521f, 0.41417847f, 0.44109412f, 0.46975890f,+ 0.50028648f, 0.53279791f, 0.56742212f, 0.60429640f,+ 0.64356699f, 0.68538959f, 0.72993007f, 0.77736504f, 0.82788260f, 0.88168307f, 0.9389798f, 1.0f }; @@ -2010,7 +2047,7 @@ ady -= abs(base) * adx; if (x1 > n) x1 = n; if (x < x1) {- LINE_OP(output[x], inverse_db_table[y]);+ LINE_OP(output[x], inverse_db_table[y&255]); for (++x; x < x1; ++x) { err += ady; if (err >= adx) {@@ -2018,7 +2055,7 @@ y += sy; } else y += base;- LINE_OP(output[x], inverse_db_table[y]);+ LINE_OP(output[x], inverse_db_table[y&255]); } } }@@ -2124,47 +2161,7 @@ ++class_set; #endif }- } else if (ch == 1) {- while (pcount < part_read) {- int z = r->begin + pcount*r->part_size;- int c_inter = 0, p_inter = z;- if (pass == 0) {- Codebook *c = f->codebooks+r->classbook;- int q;- DECODE(q,f,c);- if (q == EOP) goto done;- #ifndef STB_VORBIS_DIVIDES_IN_RESIDUE- part_classdata[0][class_set] = r->classdata[q];- #else- for (i=classwords-1; i >= 0; --i) {- classifications[0][i+pcount] = q % r->classifications;- q /= r->classifications;- }- #endif- }- for (i=0; i < classwords && pcount < part_read; ++i, ++pcount) {- int z = r->begin + pcount*r->part_size;- #ifndef STB_VORBIS_DIVIDES_IN_RESIDUE- int c = part_classdata[0][class_set][i];- #else- int c = classifications[0][pcount];- #endif- int b = r->residue_books[c][pass];- if (b >= 0) {- Codebook *book = f->codebooks + b;- if (!codebook_decode_deinterleave_repeat(f, book, residue_buffers, ch, &c_inter, &p_inter, n, r->part_size))- goto done;- } else {- z += r->part_size;- c_inter = 0;- p_inter = z;- }- }- #ifndef STB_VORBIS_DIVIDES_IN_RESIDUE- ++class_set;- #endif- }- } else {+ } else if (ch > 2) { while (pcount < part_read) { int z = r->begin + pcount*r->part_size; int c_inter = z % ch, p_inter = z/ch;@@ -2351,11 +2348,11 @@ #if LIBVORBIS_MDCT // directly call the vorbis MDCT using an interface documented // by Jeff Roberts... useful for performance comparison-typedef struct +typedef struct { int n; int log2n;- + float *trig; int *bitrev; @@ -2374,7 +2371,7 @@ if (M1.n == n) M = &M1; else if (M2.n == n) M = &M2; else if (M1.n == 0) { mdct_init(&M1, n); M = &M1; }- else { + else { if (M2.n) __asm int 3; mdct_init(&M2, n); M = &M2;@@ -2637,7 +2634,7 @@ // once I combined the passes. // so there's a missing 'times 2' here (for adding X to itself).- // this propogates through linearly to the end, where the numbers+ // this propagates through linearly to the end, where the numbers // are 1/2 too small, and need to be compensated for. {@@ -2787,7 +2784,7 @@ d1[0] = u[k4+1]; d0[1] = u[k4+2]; d0[0] = u[k4+3];- + d0 -= 4; d1 -= 4; bitrev += 2;@@ -2868,7 +2865,7 @@ float p0,p1,p2,p3; p3 = e[6]*B[7] - e[7]*B[6];- p2 = -e[6]*B[6] - e[7]*B[7]; + p2 = -e[6]*B[6] - e[7]*B[7]; d0[0] = p3; d1[3] = - p3;@@ -2876,7 +2873,7 @@ d3[3] = p2; p1 = e[4]*B[5] - e[5]*B[4];- p0 = -e[4]*B[4] - e[5]*B[5]; + p0 = -e[4]*B[4] - e[5]*B[5]; d0[1] = p1; d1[2] = - p1;@@ -2884,7 +2881,7 @@ d3[2] = p0; p3 = e[2]*B[3] - e[3]*B[2];- p2 = -e[2]*B[2] - e[3]*B[3]; + p2 = -e[2]*B[2] - e[3]*B[3]; d0[2] = p3; d1[1] = - p3;@@ -2892,7 +2889,7 @@ d3[1] = p2; p1 = e[0]*B[1] - e[1]*B[0];- p0 = -e[0]*B[0] - e[1]*B[1]; + p0 = -e[0]*B[0] - e[1]*B[1]; d0[3] = p1; d1[0] = - p1;@@ -3045,7 +3042,6 @@ len <<= 1; if (len == f->blocksize_0) return f->window[0]; if (len == f->blocksize_1) return f->window[1];- assert(0); return NULL; } @@ -3451,6 +3447,7 @@ if (f->previous_length) { int i,j, n = f->previous_length; float *w = get_window(f, n);+ if (w == NULL) return 0; for (i=0; i < f->channels; ++i) { for (j=0; j < n; ++j) f->channel_buffers[i][left+j] =@@ -3498,7 +3495,7 @@ } #ifndef STB_VORBIS_NO_PUSHDATA_API-static int is_whole_packet_present(stb_vorbis *f, int end_page)+static int is_whole_packet_present(stb_vorbis *f) { // make sure that we have the packet available before continuing... // this requires a full ogg parse, but we know we can fetch from f->stream@@ -3518,15 +3515,13 @@ break; } // either this continues, or it ends it...- if (end_page)- if (s < f->segment_count-1) return error(f, VORBIS_invalid_stream); if (s == f->segment_count) s = -1; // set 'crosses page' flag if (p > f->stream_end) return error(f, VORBIS_need_more_data); first = FALSE; } for (; s == -1;) {- uint8 *q; + uint8 *q; int n; // check that we have the page header ready@@ -3552,8 +3547,6 @@ if (q[s] < 255) break; }- if (end_page)- if (s < n-1) return error(f, VORBIS_invalid_stream); if (s == n) s = -1; // set 'crosses page' flag if (p > f->stream_end) return error(f, VORBIS_need_more_data);@@ -3570,6 +3563,7 @@ int longest_floorlist=0; // first page, first packet+ f->first_decode = TRUE; if (!start_page(f)) return FALSE; // validate page flag@@ -3578,7 +3572,22 @@ if (f->page_flag & PAGEFLAG_continued_packet) return error(f, VORBIS_invalid_first_page); // check for expected packet length if (f->segment_count != 1) return error(f, VORBIS_invalid_first_page);- if (f->segments[0] != 30) return error(f, VORBIS_invalid_first_page);+ if (f->segments[0] != 30) {+ // check for the Ogg skeleton fishead identifying header to refine our error+ if (f->segments[0] == 64 &&+ getn(f, header, 6) &&+ header[0] == 'f' &&+ header[1] == 'i' &&+ header[2] == 's' &&+ header[3] == 'h' &&+ header[4] == 'e' &&+ header[5] == 'a' &&+ get8(f) == 'd' &&+ get8(f) == '\0') return error(f, VORBIS_ogg_skeleton_not_supported);+ else+ return error(f, VORBIS_invalid_first_page);+ }+ // read packet // check packet header if (get8(f) != VORBIS_packet_id) return error(f, VORBIS_invalid_first_page);@@ -3612,6 +3621,41 @@ if (!start_page(f)) return FALSE; if (!start_packet(f)) return FALSE;++ if (!next_segment(f)) return FALSE;++ if (get8_packet(f) != VORBIS_packet_comment) return error(f, VORBIS_invalid_setup);+ for (i=0; i < 6; ++i) header[i] = get8_packet(f);+ if (!vorbis_validate(header)) return error(f, VORBIS_invalid_setup);+ //file vendor+ len = get32_packet(f);+ f->vendor = (char*)setup_malloc(f, sizeof(char) * (len+1));+ for(i=0; i < len; ++i) {+ f->vendor[i] = get8_packet(f);+ }+ f->vendor[len] = (char)'\0';+ //user comments+ f->comment_list_length = get32_packet(f);+ f->comment_list = (char**)setup_malloc(f, sizeof(char*) * (f->comment_list_length));++ for(i=0; i < f->comment_list_length; ++i) {+ len = get32_packet(f);+ f->comment_list[i] = (char*)setup_malloc(f, sizeof(char) * (len+1));++ for(j=0; j < len; ++j) {+ f->comment_list[i][j] = get8_packet(f);+ }+ f->comment_list[i][len] = (char)'\0';+ }++ // framing_flag+ x = get8_packet(f);+ if (!(x & 1)) return error(f, VORBIS_invalid_setup);+++ skip(f, f->bytes_in_seg);+ f->bytes_in_seg = 0;+ do { len = next_segment(f); skip(f, len);@@ -3623,7 +3667,7 @@ #ifndef STB_VORBIS_NO_PUSHDATA_API if (IS_PUSH_MODE(f)) {- if (!is_whole_packet_present(f, TRUE)) {+ if (!is_whole_packet_present(f)) { // convert error in ogg header to write type if (f->error == VORBIS_invalid_stream) f->error = VORBIS_invalid_setup;@@ -3677,6 +3721,7 @@ while (current_entry < c->entries) { int limit = c->entries - current_entry; int n = get_bits(f, ilog(limit));+ if (current_length >= 32) return error(f, VORBIS_invalid_setup); if (current_entry + n > (int) c->entries) { return error(f, VORBIS_invalid_setup); } memset(lengths + current_entry, current_length, n); current_entry += n;@@ -3780,7 +3825,9 @@ c->value_bits = get_bits(f, 4)+1; c->sequence_p = get_bits(f,1); if (c->lookup_type == 1) {- c->lookup_values = lookup1_values(c->entries, c->dimensions);+ int values = lookup1_values(c->entries, c->dimensions);+ if (values < 0) return error(f, VORBIS_invalid_setup);+ c->lookup_values = (uint32) values; } else { c->lookup_values = c->entries * c->dimensions; }@@ -3879,7 +3926,7 @@ } else { stbv__floor_ordering p[31*8+2]; Floor1 *g = &f->floor_config[i].floor1;- int max_class = -1; + int max_class = -1; g->partitions = get_bits(f, 5); for (j=0; j < g->partitions; ++j) { g->partition_class_list[j] = get_bits(f, 4);@@ -3916,11 +3963,14 @@ p[j].id = j; } qsort(p, g->values, sizeof(p[0]), point_compare);+ for (j=0; j < g->values-1; ++j)+ if (p[j].x == p[j+1].x)+ return error(f, VORBIS_invalid_setup); for (j=0; j < g->values; ++j) g->sorted_order[j] = (uint8) p[j].id; // precompute the neighbors for (j=2; j < g->values; ++j) {- int low,hi;+ int low = 0,hi = 0; neighbors(g->Xlist, j, &low,&hi); g->neighbors[j][0] = low; g->neighbors[j][1] = hi;@@ -3989,7 +4039,7 @@ if (f->mapping == NULL) return error(f, VORBIS_outofmem); memset(f->mapping, 0, f->mapping_count * sizeof(*f->mapping)); for (i=0; i < f->mapping_count; ++i) {- Mapping *m = f->mapping + i; + Mapping *m = f->mapping + i; int mapping_type = get_bits(f,16); if (mapping_type != 0) return error(f, VORBIS_invalid_setup); m->chan = (MappingChannel *) setup_malloc(f, f->channels * sizeof(*m->chan));@@ -4002,6 +4052,7 @@ max_submaps = m->submaps; if (get_bits(f,1)) { m->coupling_steps = get_bits(f,8)+1;+ if (m->coupling_steps > f->channels) return error(f, VORBIS_invalid_setup); for (k=0; k < m->coupling_steps; ++k) { m->chan[k].magnitude = get_bits(f, ilog(f->channels-1)); m->chan[k].angle = get_bits(f, ilog(f->channels-1));@@ -4104,7 +4155,6 @@ f->temp_memory_required = imdct_mem; } - f->first_decode = TRUE; if (f->alloc.alloc_buffer) { assert(f->temp_offset == f->alloc.alloc_buffer_length_in_bytes);@@ -4113,7 +4163,17 @@ return error(f, VORBIS_outofmem); } - f->first_audio_page_offset = stb_vorbis_get_file_offset(f);+ // @TODO: stb_vorbis_seek_start expects first_audio_page_offset to point to a page+ // without PAGEFLAG_continued_packet, so this either points to the first page, or+ // the page after the end of the headers. It might be cleaner to point to a page+ // in the middle of the headers, when that's the page where the first audio packet+ // starts, but we'd have to also correctly skip the end of any continued packet in+ // stb_vorbis_seek_start.+ if (f->next_seg == -1) {+ f->first_audio_page_offset = stb_vorbis_get_file_offset(f);+ } else {+ f->first_audio_page_offset = 0;+ } return TRUE; }@@ -4121,6 +4181,13 @@ static void vorbis_deinit(stb_vorbis *p) { int i,j;++ setup_free(p, p->vendor);+ for (i=0; i < p->comment_list_length; ++i) {+ setup_free(p, p->comment_list[i]);+ }+ setup_free(p, p->comment_list);+ if (p->residue_config) { for (i=0; i < p->residue_count; ++i) { Residue *r = p->residue_config+i;@@ -4220,6 +4287,15 @@ return d; } +stb_vorbis_comment stb_vorbis_get_comment(stb_vorbis *f)+{+ stb_vorbis_comment d;+ d.vendor = f->vendor;+ d.comment_list_length = f->comment_list_length;+ d.comment_list = f->comment_list;+ return d;+}+ int stb_vorbis_get_error(stb_vorbis *f) { int e = f->error;@@ -4361,7 +4437,7 @@ f->error = VORBIS__no_error; // check that we have the entire packet in memory- if (!is_whole_packet_present(f, FALSE)) {+ if (!is_whole_packet_present(f)) { *samples = 0; return 0; }@@ -4566,7 +4642,7 @@ return 1; } -// rarely used function to seek back to the preceeding page while finding the+// rarely used function to seek back to the preceding page while finding the // start of a packet static int go_to_page_before(stb_vorbis *f, unsigned int limit_offset) {@@ -4597,8 +4673,8 @@ { ProbedPage left, right, mid; int i, start_seg_with_known_loc, end_pos, page_start;- uint32 delta, stream_length, padding;- double offset, bytes_per_sample;+ uint32 delta, stream_length, padding, last_sample_limit;+ double offset = 0.0, bytes_per_sample = 0.0; int probe = 0; // find the last page and validate the target sample@@ -4611,9 +4687,9 @@ // indicates should be the granule position (give or take one)). padding = ((f->blocksize_1 - f->blocksize_0) >> 2); if (sample_number < padding)- sample_number = 0;+ last_sample_limit = 0; else- sample_number -= padding;+ last_sample_limit = sample_number - padding; left = f->p_first; while (left.last_decoded_sample == ~0U) {@@ -4626,9 +4702,12 @@ assert(right.last_decoded_sample != ~0U); // starting from the start is handled differently- if (sample_number <= left.last_decoded_sample) {- if (stb_vorbis_seek_start(f))+ if (last_sample_limit <= left.last_decoded_sample) {+ if (stb_vorbis_seek_start(f)) {+ if (f->current_loc > sample_number)+ return error(f, VORBIS_seek_failed); return 1;+ } return 0; } @@ -4645,10 +4724,10 @@ // first probe (interpolate) double data_bytes = right.page_end - left.page_start; bytes_per_sample = data_bytes / right.last_decoded_sample;- offset = left.page_start + bytes_per_sample * (sample_number - left.last_decoded_sample);+ offset = left.page_start + bytes_per_sample * (last_sample_limit - left.last_decoded_sample); } else { // second probe (try to bound the other side)- double error = ((double) sample_number - mid.last_decoded_sample) * bytes_per_sample;+ double error = ((double) last_sample_limit - mid.last_decoded_sample) * bytes_per_sample; if (error >= 0 && error < 8000) error = 8000; if (error < 0 && error > -8000) error = -8000; offset += error * 2;@@ -4679,14 +4758,16 @@ } // if we've just found the last page again then we're in a tricky file,- // and we're close enough.- if (mid.page_start == right.page_start)- break;-- if (sample_number < mid.last_decoded_sample)- right = mid;- else- left = mid;+ // and we're close enough (if it wasn't an interpolation probe).+ if (mid.page_start == right.page_start) {+ if (probe >= 2 || delta <= 65536)+ break;+ } else {+ if (last_sample_limit < mid.last_decoded_sample)+ right = mid;+ else+ left = mid;+ } ++probe; }@@ -4802,8 +4883,8 @@ flush_packet(f); } }- // the next frame will start with the sample- assert(f->current_loc == sample_number);+ // the next frame should start with the sample+ if (f->current_loc != sample_number) return error(f, VORBIS_seek_failed); return 1; } @@ -4973,8 +5054,14 @@ stb_vorbis * stb_vorbis_open_filename(const char *filename, int *error, const stb_vorbis_alloc *alloc) {- FILE *f = fopen(filename, "rb");- if (f) + FILE *f;+#if defined(_WIN32) && defined(__STDC_WANT_SECURE_LIB__)+ if (0 != fopen_s(&f, filename, "rb"))+ f = NULL;+#else+ f = fopen(filename, "rb");+#endif+ if (f) return stb_vorbis_open_file(f, TRUE, error, alloc); if (error) *error = VORBIS_file_open_failure; return NULL;@@ -5037,7 +5124,7 @@ #define MAGIC(SHIFT) (1.5f * (1 << (23-SHIFT)) + 0.5f/(1 << SHIFT)) #define ADDEND(SHIFT) (((150-SHIFT) << 23) + (1 << 22)) #define FAST_SCALED_FLOAT_TO_INT(temp,x,s) (temp.f = (x) + MAGIC(s), temp.i - ADDEND(s))- #define check_endianness() + #define check_endianness() #else #define FAST_SCALED_FLOAT_TO_INT(temp,x,s) ((int) ((x) * (1 << (s)))) #define check_endianness()@@ -5139,7 +5226,7 @@ int stb_vorbis_get_frame_short(stb_vorbis *f, int num_c, short **buffer, int num_samples) {- float **output;+ float **output = NULL; int len = stb_vorbis_get_frame_float(f, NULL, &output); if (len > num_samples) len = num_samples; if (len)@@ -5362,14 +5449,20 @@ #endif // STB_VORBIS_NO_PULLDATA_API /* Version history+ 1.17 - 2019-07-08 - fix CVE-2019-13217, -13218, -13219, -13220, -13221, -13222, -13223+ found with Mayhem by ForAllSecure+ 1.16 - 2019-03-04 - fix warnings+ 1.15 - 2019-02-07 - explicit failure if Ogg Skeleton data is found+ 1.14 - 2018-02-11 - delete bogus dealloca usage+ 1.13 - 2018-01-29 - fix truncation of last frame (hopefully) 1.12 - 2017-11-21 - limit residue begin/end to blocksize/2 to avoid large temp allocs in bad/corrupt files- 1.11 - 2017-07-23 - fix MinGW compilation + 1.11 - 2017-07-23 - fix MinGW compilation 1.10 - 2017-03-03 - more robust seeking; fix negative ilog(); clear error in open_memory 1.09 - 2016-04-04 - back out 'avoid discarding last frame' fix from previous version 1.08 - 2016-04-02 - fixed multiple warnings; fix setup memory leaks; avoid discarding last frame of audio data 1.07 - 2015-01-16 - fixed some warnings, fix mingw, const-correct API- some more crash fixes when out of memory or with corrupt files + some more crash fixes when out of memory or with corrupt files 1.06 - 2015-08-31 - full, correct support for seeking API (Dougall Johnson) some crash fixes when out of memory or with corrupt files 1.05 - 2015-04-19 - don't define __forceinline if it's redundant@@ -5425,38 +5518,38 @@ ------------------------------------------------------------------------------ ALTERNATIVE A - MIT License Copyright (c) 2017 Sean Barrett-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 +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 +The above copyright notice and this permission notice shall be included in all copies or substantial portions of the Software.-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 +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. ------------------------------------------------------------------------------ ALTERNATIVE B - Public Domain (www.unlicense.org) This is free and unencumbered software released into the public domain.-Anyone is free to copy, modify, publish, use, compile, sell, or distribute this -software, either in source code form or as a compiled binary, for any purpose, +Anyone is free to copy, modify, publish, use, compile, sell, or distribute this+software, either in source code form or as a compiled binary, for any purpose, commercial or non-commercial, and by any means.-In jurisdictions that recognize copyright laws, the author or authors of this -software dedicate any and all copyright interest in the software to the public -domain. We make this dedication for the benefit of the public at large and to -the detriment of our heirs and successors. We intend this dedication to be an -overt act of relinquishment in perpetuity of all present and future rights to +In jurisdictions that recognize copyright laws, the author or authors of this+software dedicate any and all copyright interest in the software to the public+domain. We make this dedication for the benefit of the public at large and to+the detriment of our heirs and successors. We intend this dedication to be an+overt act of relinquishment in perpetuity of all present and future rights to this software under copyright law.-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 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 +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 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. ------------------------------------------------------------------------------ */
example/play.hs view
@@ -6,44 +6,50 @@ import Sound.ProteaAudio -waitPayback = do- n <- soundActive+waitPlayback = do+ n <- soundActiveAll when (n > 0) $ do threadDelay oneSec- waitPayback+ waitPlayback oneSec :: Int oneSec = 1000000 -- micro seconds main = do- args <- getArgs- filename <- case args of- a : _ -> pure a- _ -> fail "usage: proteaaudio-play SAMPLE_FILE_NAME"+ args <- getArgs+ filename <- case args of+ a : _ -> pure a+ _ -> fail "usage: proteaaudio-play SAMPLE_FILE_NAME" - result <- initAudio 64 44100 1024 -- max channels, mixing frequency, mixing buffer size- unless result $ fail "failed to initialize the audio system"+ result <- initAudio 64 44100 1024 -- max channels, mixing frequency, mixing buffer size+ unless result $ fail "failed to initialize the audio system" - -- (A) load sample from file- sampleA <- sampleFromFile filename 1.0 -- volume+ -- (A) load sample from file+ sampleA <- sampleFromFile filename 1.0 -- volume - -- start two sound tracks with shared sample data- sndTrkA <- soundPlay sampleA 1 1 0 1 -- left volume, right volume, time difference between left and right, pitch factor for playback- threadDelay oneSec -- wait 1 sec- sndTrkB <- soundPlay sampleA 1 1 0 1 -- left volume, right volume, time difference between left and right, pitch factor for playback- -- play 3 sec- threadDelay $ 3 * oneSec- soundStop sndTrkB- -- wait sndTrkA to finish- waitPayback+ -- start two sound tracks with shared sample data+ sndTrkA <- soundPlay sampleA 1 1 0 1 -- left volume, right volume, time difference between left and right, pitch factor for playback+ threadDelay oneSec -- wait 1 sec+ sndTrkB <- soundPlay sampleA 1 1 0 1 -- left volume, right volume, time difference between left and right, pitch factor for playback+ soundActive sndTrkB >>= print+ -- play 3 sec+ threadDelay $ 3 * oneSec+ soundStop sndTrkB+ soundActive sndTrkB >>= print+ -- wait sndTrkA to finish+ waitPlayback - -- (B) load from memory buffer- buffer <- SB.readFile filename- sampleB <- case takeExtension filename of- ".ogg" -> sampleFromMemoryOgg buffer 1.0- ".wav" -> sampleFromMemoryWav buffer 1.0+ -- (B) load from memory buffer+ buffer <- SB.readFile filename+ sampleB <- case takeExtension filename of+ ".ogg" -> sampleFromMemoryOgg buffer 1.0+ ".wav" -> sampleFromMemoryWav buffer 1.0 - soundPlay sampleB 1 1 0 1 -- left volume, right volume, time difference between left and right, pitch factor for playback- waitPayback+ soundPlay sampleB 1 1 0 1 -- left volume, right volume, time difference between left and right, pitch factor for playback+ waitPlayback - finishAudio+ sampleDestroy sampleB+ soundPlay sampleB 1 1 0 1 -- we have invalidated the handle; nothing should happen now+ waitPlayback++ finishAudio
proteaaudio.cabal view
@@ -1,5 +1,5 @@ Name: proteaaudio-Version: 0.8.0+Version: 0.9.0 Synopsis: Simple audio library for Windows, Linux, OSX. Description: Simple audio library for Windows, Linux, OSX. Supports PCM, Ogg and Wav playback and multichannel mixing. License: BSD3@@ -8,7 +8,7 @@ Maintainer: csaba.hruska@gmail.com Stability: Experimental Category: Sound-Tested-With: GHC == 8.4.4+Tested-With: GHC == 8.8.3 Cabal-Version: >= 1.10 Build-Type: Simple @@ -29,7 +29,6 @@ cbits/include/soundcard.h cbits/RtAudio.cpp cbits/RtAudio.h- cbits/RtError.h cbits/proAudio.cpp cbits/proAudio.h cbits/proAudioRt.cpp@@ -44,6 +43,7 @@ README.md CHANGELOG.md+ stack.yaml Flag example Description: Build with example@@ -71,12 +71,15 @@ Include-Dirs: cbits cbits/include + CC-Options: "-DPROTEAAUDIO_RT"+ if os(windows) CC-Options: "-D__WINDOWS_DS__" Extra-Libraries: stdc++ ole32 dsound winmm if os(linux) CC-Options: "-D__LINUX_PULSE__" Extra-Libraries: stdc++ pthread pulse-simple pulse+ pkgconfig-depends: libpulse-simple, libpulse if os(darwin) CC-Options: "-D__MACOSX_CORE__" Extra-Libraries: stdc++ pthread
+ stack.yaml view
@@ -0,0 +1,13 @@+resolver: lts-15.6++packages:+- '.'++# Override default flag values for local packages and extra-deps+flags:+ proteaaudio:+ example: true+nix:+ enable: false+ packages: [pulseaudio]+ pure: false