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@@ -1,4 +1,5 @@
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-#ifdef RTAUDIO_ENABLED
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+#ifdef RTAUDIO_ENABLED // -GODOT-
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+
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/************************************************************************/
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/*! \class RtAudio
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\brief Realtime audio i/o C++ classes.
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@@ -11,7 +12,7 @@
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RtAudio WWW site: http://www.music.mcgill.ca/~gary/rtaudio/
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RtAudio: realtime audio i/o C++ classes
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- Copyright (c) 2001-2014 Gary P. Scavone
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+ Copyright (c) 2001-2016 Gary P. Scavone
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Permission is hereby granted, free of charge, to any person
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obtaining a copy of this software and associated documentation files
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@@ -39,7 +40,7 @@
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*/
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/************************************************************************/
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-// RtAudio: Version 4.1.1
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+// RtAudio: Version 4.1.2
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#include "RtAudio.h"
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#include <iostream>
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@@ -56,11 +57,7 @@ const unsigned int RtApi::SAMPLE_RATES[] = {
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};
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#if defined(__WINDOWS_DS__) || defined(__WINDOWS_ASIO__) || defined(__WINDOWS_WASAPI__)
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-#ifdef WINRT_ENABLED
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- #define MUTEX_INITIALIZE(A) InitializeCriticalSectionEx(A, 0, 0)
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-#else
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#define MUTEX_INITIALIZE(A) InitializeCriticalSection(A)
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-#endif
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#define MUTEX_DESTROY(A) DeleteCriticalSection(A)
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#define MUTEX_LOCK(A) EnterCriticalSection(A)
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#define MUTEX_UNLOCK(A) LeaveCriticalSection(A)
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@@ -139,7 +136,8 @@ void RtAudio :: getCompiledApi( std::vector<RtAudio::Api> &apis ) throw()
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void RtAudio :: openRtApi( RtAudio::Api api )
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{
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- delete rtapi_;
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+ if ( rtapi_ )
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+ delete rtapi_;
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rtapi_ = 0;
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#if defined(__UNIX_JACK__)
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@@ -215,7 +213,8 @@ RtAudio :: RtAudio( RtAudio::Api api )
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RtAudio :: ~RtAudio() throw()
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{
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- delete rtapi_;
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+ if ( rtapi_ )
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+ delete rtapi_;
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}
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void RtAudio :: openStream( RtAudio::StreamParameters *outputParameters,
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@@ -418,7 +417,7 @@ double RtApi :: getStreamTime( void )
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then = stream_.lastTickTimestamp;
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return stream_.streamTime +
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((now.tv_sec + 0.000001 * now.tv_usec) -
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- (then.tv_sec + 0.000001 * then.tv_usec));
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+ (then.tv_sec + 0.000001 * then.tv_usec));
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#else
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return stream_.streamTime;
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#endif
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@@ -1832,7 +1831,7 @@ bool RtApiCore :: callbackEvent( AudioDeviceID deviceId,
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channelsLeft -= streamChannels;
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}
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}
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-
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+
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if ( stream_.doConvertBuffer[1] ) { // convert from our internal "device" buffer
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convertBuffer( stream_.userBuffer[1],
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stream_.deviceBuffer,
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@@ -2719,7 +2718,7 @@ RtApiAsio :: RtApiAsio()
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// CoInitialize beforehand, but it must be for appartment threading
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// (in which case, CoInitilialize will return S_FALSE here).
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coInitialized_ = false;
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- HRESULT hr = CoInitialize( NULL );
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+ HRESULT hr = CoInitialize( NULL );
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if ( FAILED(hr) ) {
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errorText_ = "RtApiAsio::ASIO requires a single-threaded appartment. Call CoInitializeEx(0,COINIT_APARTMENTTHREADED)";
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error( RtAudioError::WARNING );
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@@ -3170,7 +3169,7 @@ bool RtApiAsio :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigne
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errorText_ = errorStream_.str();
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goto error;
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}
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- buffersAllocated = true;
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+ buffersAllocated = true;
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stream_.state = STREAM_STOPPED;
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// Set flags for buffer conversion.
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@@ -3644,13 +3643,13 @@ static long asioMessages( long selector, long value, void* /*message*/, double*
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static const char* getAsioErrorString( ASIOError result )
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{
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- struct Messages
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+ struct Messages
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{
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ASIOError value;
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const char*message;
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};
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- static const Messages m[] =
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+ static const Messages m[] =
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{
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{ ASE_NotPresent, "Hardware input or output is not present or available." },
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{ ASE_HWMalfunction, "Hardware is malfunctioning." },
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@@ -5161,10 +5160,10 @@ void RtApiWasapi::wasapiThread()
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// if the callback buffer was pushed renderBuffer reset callbackPulled flag
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if ( callbackPushed ) {
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callbackPulled = false;
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+ // tick stream time
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+ RtApi::tickStreamTime();
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}
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- // tick stream time
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- RtApi::tickStreamTime();
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}
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Exit:
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@@ -5192,7 +5191,7 @@ Exit:
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#if defined(__WINDOWS_DS__) // Windows DirectSound API
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// Modified by Robin Davies, October 2005
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-// - Improvements to DirectX pointer chasing.
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+// - Improvements to DirectX pointer chasing.
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// - Bug fix for non-power-of-two Asio granularity used by Edirol PCR-A30.
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// - Auto-call CoInitialize for DSOUND and ASIO platforms.
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// Various revisions for RtAudio 4.0 by Gary Scavone, April 2007
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@@ -5232,7 +5231,7 @@ struct DsHandle {
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void *id[2];
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void *buffer[2];
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bool xrun[2];
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- UINT bufferPointer[2];
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+ UINT bufferPointer[2];
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DWORD dsBufferSize[2];
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DWORD dsPointerLeadTime[2]; // the number of bytes ahead of the safe pointer to lead by.
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HANDLE condition;
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@@ -6080,7 +6079,7 @@ void RtApiDs :: startStream()
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// Increase scheduler frequency on lesser windows (a side-effect of
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// increasing timer accuracy). On greater windows (Win2K or later),
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// this is already in effect.
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- timeBeginPeriod( 1 );
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+ timeBeginPeriod( 1 );
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buffersRolling = false;
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duplexPrerollBytes = 0;
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@@ -6401,7 +6400,7 @@ void RtApiDs :: callbackEvent()
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}
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if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
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-
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+
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LPDIRECTSOUNDBUFFER dsBuffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0];
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if ( handle->drainCounter > 1 ) { // write zeros to the output stream
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@@ -6441,6 +6440,7 @@ void RtApiDs :: callbackEvent()
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if ( FAILED( result ) ) {
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errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current write position!";
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errorText_ = errorStream_.str();
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+ MUTEX_UNLOCK( &stream_.mutex );
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error( RtAudioError::SYSTEM_ERROR );
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return;
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}
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@@ -6467,7 +6467,7 @@ void RtApiDs :: callbackEvent()
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}
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if ( dsPointerBetween( nextWritePointer, safeWritePointer, currentWritePointer, dsBufferSize )
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- || dsPointerBetween( endWrite, safeWritePointer, currentWritePointer, dsBufferSize ) ) {
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+ || dsPointerBetween( endWrite, safeWritePointer, currentWritePointer, dsBufferSize ) ) {
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// We've strayed into the forbidden zone ... resync the read pointer.
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handle->xrun[0] = true;
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nextWritePointer = safeWritePointer + handle->dsPointerLeadTime[0] - bufferBytes;
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@@ -6541,14 +6541,14 @@ void RtApiDs :: callbackEvent()
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if ( safeReadPointer < (DWORD)nextReadPointer ) safeReadPointer += dsBufferSize; // unwrap offset
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DWORD endRead = nextReadPointer + bufferBytes;
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- // Handling depends on whether we are INPUT or DUPLEX.
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+ // Handling depends on whether we are INPUT or DUPLEX.
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// If we're in INPUT mode then waiting is a good thing. If we're in DUPLEX mode,
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// then a wait here will drag the write pointers into the forbidden zone.
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- //
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- // In DUPLEX mode, rather than wait, we will back off the read pointer until
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- // it's in a safe position. This causes dropouts, but it seems to be the only
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- // practical way to sync up the read and write pointers reliably, given the
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- // the very complex relationship between phase and increment of the read and write
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+ //
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+ // In DUPLEX mode, rather than wait, we will back off the read pointer until
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+ // it's in a safe position. This causes dropouts, but it seems to be the only
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+ // practical way to sync up the read and write pointers reliably, given the
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+ // the very complex relationship between phase and increment of the read and write
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// pointers.
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//
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// In order to minimize audible dropouts in DUPLEX mode, we will
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@@ -6599,7 +6599,7 @@ void RtApiDs :: callbackEvent()
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error( RtAudioError::SYSTEM_ERROR );
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return;
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}
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-
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+
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if ( safeReadPointer < (DWORD)nextReadPointer ) safeReadPointer += dsBufferSize; // unwrap offset
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}
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}
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@@ -7809,7 +7809,7 @@ void RtApiAlsa :: stopStream()
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AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle;
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snd_pcm_t **handle = (snd_pcm_t **) apiInfo->handles;
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if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
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- if ( apiInfo->synchronized )
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+ if ( apiInfo->synchronized )
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result = snd_pcm_drop( handle[0] );
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else
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result = snd_pcm_drain( handle[0] );
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@@ -8073,7 +8073,7 @@ static void *alsaCallbackHandler( void *ptr )
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bool *isRunning = &info->isRunning;
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#ifdef SCHED_RR // Undefined with some OSes (eg: NetBSD 1.6.x with GNU Pthread)
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- if ( &info->doRealtime ) {
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+ if ( info->doRealtime ) {
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pthread_t tID = pthread_self(); // ID of this thread
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sched_param prio = { info->priority }; // scheduling priority of thread
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pthread_setschedparam( tID, SCHED_RR, &prio );
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@@ -8277,7 +8277,7 @@ void RtApiPulse::callbackEvent( void )
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else
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bytes = stream_.nUserChannels[INPUT] * stream_.bufferSize *
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formatBytes( stream_.userFormat );
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-
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+
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if ( pa_simple_read( pah->s_rec, pulse_in, bytes, &pa_error ) < 0 ) {
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errorStream_ << "RtApiPulse::callbackEvent: audio read error, " <<
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pa_strerror( pa_error ) << ".";
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@@ -8525,7 +8525,7 @@ bool RtApiPulse::probeDeviceOpen( unsigned int device, StreamMode mode,
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}
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break;
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case OUTPUT:
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- pah->s_play = pa_simple_new( NULL, "RtAudio", PA_STREAM_PLAYBACK, NULL, "Playback", &ss, NULL, NULL, &error );
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+ pah->s_play = pa_simple_new( NULL, streamName.c_str(), PA_STREAM_PLAYBACK, NULL, "Playback", &ss, NULL, NULL, &error );
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if ( !pah->s_play ) {
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errorText_ = "RtApiPulse::probeDeviceOpen: error connecting output to PulseAudio server.";
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goto error;
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@@ -8553,7 +8553,7 @@ bool RtApiPulse::probeDeviceOpen( unsigned int device, StreamMode mode,
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stream_.state = STREAM_STOPPED;
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return true;
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-
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+
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error:
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if ( pah && stream_.callbackInfo.isRunning ) {
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pthread_cond_destroy( &pah->runnable_cv );
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@@ -10145,8 +10145,8 @@ void RtApi :: convertBuffer( char *outBuffer, char *inBuffer, ConvertInfo &info
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void RtApi :: byteSwapBuffer( char *buffer, unsigned int samples, RtAudioFormat format )
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{
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- register char val;
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- register char *ptr;
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+ char val;
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+ char *ptr;
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ptr = buffer;
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if ( format == RTAUDIO_SINT16 ) {
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@@ -10229,4 +10229,4 @@ void RtApi :: byteSwapBuffer( char *buffer, unsigned int samples, RtAudioFormat
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//
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// vim: et sts=2 sw=2
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-#endif
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+#endif // RTAUDIO_ENABLED -GODOT-
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Rémi Verschelde