diff options
Diffstat (limited to 'drivers/rtaudio/RtAudio.cpp')
| -rw-r--r-- | drivers/rtaudio/RtAudio.cpp | 74 |
1 files changed, 37 insertions, 37 deletions
diff --git a/drivers/rtaudio/RtAudio.cpp b/drivers/rtaudio/RtAudio.cpp index 86981e56d2..04159776f7 100644 --- a/drivers/rtaudio/RtAudio.cpp +++ b/drivers/rtaudio/RtAudio.cpp @@ -1,4 +1,5 @@ -#ifdef RTAUDIO_ENABLED +#ifdef RTAUDIO_ENABLED // -GODOT- + /************************************************************************/ /*! \class RtAudio \brief Realtime audio i/o C++ classes. @@ -11,7 +12,7 @@ RtAudio WWW site: http://www.music.mcgill.ca/~gary/rtaudio/ RtAudio: realtime audio i/o C++ classes - Copyright (c) 2001-2014 Gary P. Scavone + Copyright (c) 2001-2016 Gary P. Scavone Permission is hereby granted, free of charge, to any person obtaining a copy of this software and associated documentation files @@ -39,7 +40,7 @@ */ /************************************************************************/ -// RtAudio: Version 4.1.1 +// RtAudio: Version 4.1.2 #include "RtAudio.h" #include <iostream> @@ -56,11 +57,7 @@ const unsigned int RtApi::SAMPLE_RATES[] = { }; #if defined(__WINDOWS_DS__) || defined(__WINDOWS_ASIO__) || defined(__WINDOWS_WASAPI__) -#ifdef WINRT_ENABLED - #define MUTEX_INITIALIZE(A) InitializeCriticalSectionEx(A, 0, 0) -#else #define MUTEX_INITIALIZE(A) InitializeCriticalSection(A) -#endif #define MUTEX_DESTROY(A) DeleteCriticalSection(A) #define MUTEX_LOCK(A) EnterCriticalSection(A) #define MUTEX_UNLOCK(A) LeaveCriticalSection(A) @@ -139,7 +136,8 @@ void RtAudio :: getCompiledApi( std::vector<RtAudio::Api> &apis ) throw() void RtAudio :: openRtApi( RtAudio::Api api ) { - delete rtapi_; + if ( rtapi_ ) + delete rtapi_; rtapi_ = 0; #if defined(__UNIX_JACK__) @@ -215,7 +213,8 @@ RtAudio :: RtAudio( RtAudio::Api api ) RtAudio :: ~RtAudio() throw() { - delete rtapi_; + if ( rtapi_ ) + delete rtapi_; } void RtAudio :: openStream( RtAudio::StreamParameters *outputParameters, @@ -418,7 +417,7 @@ double RtApi :: getStreamTime( void ) then = stream_.lastTickTimestamp; return stream_.streamTime + ((now.tv_sec + 0.000001 * now.tv_usec) - - (then.tv_sec + 0.000001 * then.tv_usec)); + (then.tv_sec + 0.000001 * then.tv_usec)); #else return stream_.streamTime; #endif @@ -1832,7 +1831,7 @@ bool RtApiCore :: callbackEvent( AudioDeviceID deviceId, channelsLeft -= streamChannels; } } - + if ( stream_.doConvertBuffer[1] ) { // convert from our internal "device" buffer convertBuffer( stream_.userBuffer[1], stream_.deviceBuffer, @@ -2719,7 +2718,7 @@ RtApiAsio :: RtApiAsio() // CoInitialize beforehand, but it must be for appartment threading // (in which case, CoInitilialize will return S_FALSE here). coInitialized_ = false; - HRESULT hr = CoInitialize( NULL ); + HRESULT hr = CoInitialize( NULL ); if ( FAILED(hr) ) { errorText_ = "RtApiAsio::ASIO requires a single-threaded appartment. Call CoInitializeEx(0,COINIT_APARTMENTTHREADED)"; error( RtAudioError::WARNING ); @@ -3170,7 +3169,7 @@ bool RtApiAsio :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigne errorText_ = errorStream_.str(); goto error; } - buffersAllocated = true; + buffersAllocated = true; stream_.state = STREAM_STOPPED; // Set flags for buffer conversion. @@ -3644,13 +3643,13 @@ static long asioMessages( long selector, long value, void* /*message*/, double* static const char* getAsioErrorString( ASIOError result ) { - struct Messages + struct Messages { ASIOError value; const char*message; }; - static const Messages m[] = + static const Messages m[] = { { ASE_NotPresent, "Hardware input or output is not present or available." }, { ASE_HWMalfunction, "Hardware is malfunctioning." }, @@ -5161,10 +5160,10 @@ void RtApiWasapi::wasapiThread() // if the callback buffer was pushed renderBuffer reset callbackPulled flag if ( callbackPushed ) { callbackPulled = false; + // tick stream time + RtApi::tickStreamTime(); } - // tick stream time - RtApi::tickStreamTime(); } Exit: @@ -5192,7 +5191,7 @@ Exit: #if defined(__WINDOWS_DS__) // Windows DirectSound API // Modified by Robin Davies, October 2005 -// - Improvements to DirectX pointer chasing. +// - 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 @@ -5232,7 +5231,7 @@ struct DsHandle { void *id[2]; void *buffer[2]; bool xrun[2]; - UINT bufferPointer[2]; + UINT bufferPointer[2]; DWORD dsBufferSize[2]; DWORD dsPointerLeadTime[2]; // the number of bytes ahead of the safe pointer to lead by. HANDLE condition; @@ -6080,7 +6079,7 @@ void RtApiDs :: startStream() // 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 ); + timeBeginPeriod( 1 ); buffersRolling = false; duplexPrerollBytes = 0; @@ -6401,7 +6400,7 @@ void RtApiDs :: callbackEvent() } if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { - + LPDIRECTSOUNDBUFFER dsBuffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0]; if ( handle->drainCounter > 1 ) { // write zeros to the output stream @@ -6441,6 +6440,7 @@ void RtApiDs :: callbackEvent() 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; } @@ -6467,7 +6467,7 @@ void RtApiDs :: callbackEvent() } if ( dsPointerBetween( nextWritePointer, safeWritePointer, currentWritePointer, dsBufferSize ) - || dsPointerBetween( endWrite, 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; @@ -6541,14 +6541,14 @@ void RtApiDs :: callbackEvent() if ( safeReadPointer < (DWORD)nextReadPointer ) safeReadPointer += dsBufferSize; // unwrap offset DWORD endRead = nextReadPointer + bufferBytes; - // Handling depends on whether we are INPUT or DUPLEX. + // 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 + // + // 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 @@ -6599,7 +6599,7 @@ void RtApiDs :: callbackEvent() error( RtAudioError::SYSTEM_ERROR ); return; } - + if ( safeReadPointer < (DWORD)nextReadPointer ) safeReadPointer += dsBufferSize; // unwrap offset } } @@ -7809,7 +7809,7 @@ void RtApiAlsa :: stopStream() AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle; snd_pcm_t **handle = (snd_pcm_t **) apiInfo->handles; if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { - if ( apiInfo->synchronized ) + if ( apiInfo->synchronized ) result = snd_pcm_drop( handle[0] ); else result = snd_pcm_drain( handle[0] ); @@ -8073,7 +8073,7 @@ static void *alsaCallbackHandler( void *ptr ) bool *isRunning = &info->isRunning; #ifdef SCHED_RR // Undefined with some OSes (eg: NetBSD 1.6.x with GNU Pthread) - if ( &info->doRealtime ) { + if ( info->doRealtime ) { pthread_t tID = pthread_self(); // ID of this thread sched_param prio = { info->priority }; // scheduling priority of thread pthread_setschedparam( tID, SCHED_RR, &prio ); @@ -8277,7 +8277,7 @@ void RtApiPulse::callbackEvent( void ) 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 ) << "."; @@ -8525,7 +8525,7 @@ bool RtApiPulse::probeDeviceOpen( unsigned int device, StreamMode mode, } break; case OUTPUT: - pah->s_play = pa_simple_new( NULL, "RtAudio", PA_STREAM_PLAYBACK, NULL, "Playback", &ss, NULL, NULL, &error ); + pah->s_play = pa_simple_new( NULL, streamName.c_str(), PA_STREAM_PLAYBACK, NULL, "Playback", &ss, NULL, NULL, &error ); if ( !pah->s_play ) { errorText_ = "RtApiPulse::probeDeviceOpen: error connecting output to PulseAudio server."; goto error; @@ -8553,7 +8553,7 @@ bool RtApiPulse::probeDeviceOpen( unsigned int device, StreamMode mode, stream_.state = STREAM_STOPPED; return true; - + error: if ( pah && stream_.callbackInfo.isRunning ) { pthread_cond_destroy( &pah->runnable_cv ); @@ -10145,8 +10145,8 @@ void RtApi :: convertBuffer( char *outBuffer, char *inBuffer, ConvertInfo &info void RtApi :: byteSwapBuffer( char *buffer, unsigned int samples, RtAudioFormat format ) { - register char val; - register char *ptr; + char val; + char *ptr; ptr = buffer; if ( format == RTAUDIO_SINT16 ) { @@ -10229,4 +10229,4 @@ void RtApi :: byteSwapBuffer( char *buffer, unsigned int samples, RtAudioFormat // // vim: et sts=2 sw=2 -#endif +#endif // RTAUDIO_ENABLED -GODOT- |