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Updated to RtAudio 4.1.1; Fixed typo in ElectricGuitar script.

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This commit is contained in:
Gary Scavone
2014-04-26 15:04:32 -04:00
parent dc2bfe01d0
commit 4e4e8053a5
3 changed files with 94 additions and 74 deletions
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10
include/RtAudio.h
View File

@@ -45,7 +45,7 @@
#ifndef __RTAUDIO_H
#define __RTAUDIO_H
#define RTAUDIO_VERSION "4.1.1pre"
#define RTAUDIO_VERSION "4.1.1"
#include <string>
#include <vector>
@@ -538,6 +538,12 @@ class RtAudio
*/
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
@@ -686,6 +692,7 @@ public:
long getStreamLatency( void );
unsigned int getStreamSampleRate( void );
virtual double getStreamTime( void );
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; }
@@ -830,6 +837,7 @@ inline bool RtAudio :: isStreamRunning( void ) const throw() { return rtapi_->is
inline long RtAudio :: getStreamLatency( void ) { return rtapi_->getStreamLatency(); }
inline unsigned int RtAudio :: getStreamSampleRate( void ) { return rtapi_->getStreamSampleRate(); }
inline double RtAudio :: getStreamTime( void ) { return rtapi_->getStreamTime(); }
inline void RtAudio :: setStreamTime( double time ) { return rtapi_->setStreamTime( time ); }
inline void RtAudio :: showWarnings( bool value ) throw() { rtapi_->showWarnings( value ); }
// RtApi Subclass prototypes.

2
projects/eguitar/ElectricGuitar
View File

@@ -1 +1 @@
wish < tcl/EGuitar.tcl | ./eguitar -or -ip
wish < tcl/Eguitar.tcl | ./eguitar -or -ip

156
src/RtAudio.cpp
View File

@@ -38,7 +38,7 @@
*/
/************************************************************************/
// RtAudio: Version 4.1.1pre
// RtAudio: Version 4.1.1
#include "RtAudio.h"
#include <iostream>
@@ -404,6 +404,14 @@ double RtApi :: getStreamTime( void )
#endif
}
void RtApi :: setStreamTime( double time )
{
verifyStream();
if ( time >= 0.0 )
stream_.streamTime = time;
}
unsigned int RtApi :: getStreamSampleRate( void )
{
verifyStream();
@@ -1682,11 +1690,12 @@ bool RtApiCore :: callbackEvent( AudioDeviceID deviceId,
}
}
}
}
if ( handle->drainCounter ) {
handle->drainCounter++;
goto unlock;
}
// Don't bother draining input
if ( handle->drainCounter ) {
handle->drainCounter++;
goto unlock;
}
AudioDeviceID inputDevice;
@@ -2572,11 +2581,12 @@ bool RtApiJack :: callbackEvent( unsigned long nframes )
memcpy( jackbuffer, &stream_.userBuffer[0][i*bufferBytes], bufferBytes );
}
}
}
if ( handle->drainCounter ) {
handle->drainCounter++;
goto unlock;
}
// Don't bother draining input
if ( handle->drainCounter ) {
handle->drainCounter++;
goto unlock;
}
if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {
@@ -3403,11 +3413,12 @@ bool RtApiAsio :: callbackEvent( long bufferIndex )
}
}
}
if ( handle->drainCounter ) {
handle->drainCounter++;
goto unlock;
}
// Don't bother draining input
if ( handle->drainCounter ) {
handle->drainCounter++;
goto unlock;
}
if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {
@@ -3582,7 +3593,7 @@ static const char* getAsioErrorString( ASIOError result )
// - 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, buffer size and channel count
// - Includes automatic internal conversion of sample rate and buffer size between hardware and the user
#ifndef INITGUID
#define INITGUID
@@ -3764,15 +3775,14 @@ private:
//-----------------------------------------------------------------------------
// In order to satisfy WASAPI's buffer requirements, we need a means of converting sample rate and
// channel counts between HW and the user. The convertBufferWasapi function is used to perform
// these conversions between HwIn->UserIn and UserOut->HwOut during the stream callback loop.
// In order to satisfy WASAPI's buffer requirements, we need a means of converting sample rate
// between HW and the user. The convertBufferWasapi function is used to perform this conversion
// between HwIn->UserIn and UserOut->HwOut during the stream callback loop.
// This sample rate converter favors speed over quality, and works best with conversions between
// one rate and its multiple.
void convertBufferWasapi( char* outBuffer,
const char* inBuffer,
const unsigned int& inChannelCount,
const unsigned int& outChannelCount,
const unsigned int& channelCount,
const unsigned int& inSampleRate,
const unsigned int& outSampleRate,
const unsigned int& inSampleCount,
@@ -3783,7 +3793,6 @@ void convertBufferWasapi( char* outBuffer,
float sampleRatio = ( float ) outSampleRate / inSampleRate;
float sampleStep = 1.0f / sampleRatio;
float inSampleFraction = 0.0f;
unsigned int commonChannelCount = std::min( inChannelCount, outChannelCount );
outSampleCount = ( unsigned int ) ( inSampleCount * sampleRatio );
@@ -3795,22 +3804,22 @@ void convertBufferWasapi( char* outBuffer,
switch ( format )
{
case RTAUDIO_SINT8:
memcpy( &( ( char* ) outBuffer )[ outSample * outChannelCount ], &( ( char* ) inBuffer )[ inSample * inChannelCount ], commonChannelCount * sizeof( char ) );
memcpy( &( ( char* ) outBuffer )[ outSample * channelCount ], &( ( char* ) inBuffer )[ inSample * channelCount ], channelCount * sizeof( char ) );
break;
case RTAUDIO_SINT16:
memcpy( &( ( short* ) outBuffer )[ outSample * outChannelCount ], &( ( short* ) inBuffer )[ inSample * inChannelCount ], commonChannelCount * sizeof( short ) );
memcpy( &( ( short* ) outBuffer )[ outSample * channelCount ], &( ( short* ) inBuffer )[ inSample * channelCount ], channelCount * sizeof( short ) );
break;
case RTAUDIO_SINT24:
memcpy( &( ( S24* ) outBuffer )[ outSample * outChannelCount ], &( ( S24* ) inBuffer )[ inSample * inChannelCount ], commonChannelCount * sizeof( S24 ) );
memcpy( &( ( S24* ) outBuffer )[ outSample * channelCount ], &( ( S24* ) inBuffer )[ inSample * channelCount ], channelCount * sizeof( S24 ) );
break;
case RTAUDIO_SINT32:
memcpy( &( ( int* ) outBuffer )[ outSample * outChannelCount ], &( ( int* ) inBuffer )[ inSample * inChannelCount ], commonChannelCount * sizeof( int ) );
memcpy( &( ( int* ) outBuffer )[ outSample * channelCount ], &( ( int* ) inBuffer )[ inSample * channelCount ], channelCount * sizeof( int ) );
break;
case RTAUDIO_FLOAT32:
memcpy( &( ( float* ) outBuffer )[ outSample * outChannelCount ], &( ( float* ) inBuffer )[ inSample * inChannelCount ], commonChannelCount * sizeof( float ) );
memcpy( &( ( float* ) outBuffer )[ outSample * channelCount ], &( ( float* ) inBuffer )[ inSample * channelCount ], channelCount * sizeof( float ) );
break;
case RTAUDIO_FLOAT64:
memcpy( &( ( double* ) outBuffer )[ outSample * outChannelCount ], &( ( double* ) inBuffer )[ inSample * inChannelCount ], commonChannelCount * sizeof( double ) );
memcpy( &( ( double* ) outBuffer )[ outSample * channelCount ], &( ( double* ) inBuffer )[ inSample * channelCount ], channelCount * sizeof( double ) );
break;
}
@@ -3847,7 +3856,6 @@ RtApiWasapi::RtApiWasapi()
{
// WASAPI can run either apartment or multi-threaded
HRESULT hr = CoInitialize( NULL );
if ( !FAILED( hr ) )
coInitialized_ = true;
@@ -3866,16 +3874,14 @@ RtApiWasapi::RtApiWasapi()
RtApiWasapi::~RtApiWasapi()
{
// if this object previously called CoInitialize()
if ( coInitialized_ ) {
CoUninitialize();
}
if ( stream_.state != STREAM_CLOSED ) {
if ( stream_.state != STREAM_CLOSED )
closeStream();
}
SAFE_RELEASE( deviceEnumerator_ );
// If this object previously called CoInitialize()
if ( coInitialized_ )
CoUninitialize();
}
//=============================================================================
@@ -4508,10 +4514,11 @@ bool RtApiWasapi::probeDeviceOpen( unsigned int device, StreamMode mode, unsigne
// Set flags for buffer conversion.
stream_.doConvertBuffer[mode] = false;
if ( stream_.userFormat != stream_.deviceFormat[mode] )
if ( stream_.userFormat != stream_.deviceFormat[mode] ||
stream_.nUserChannels != stream_.nDeviceChannels )
stream_.doConvertBuffer[mode] = true;
if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] &&
stream_.nUserChannels[mode] > 1 )
else if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] &&
stream_.nUserChannels[mode] > 1 )
stream_.doConvertBuffer[mode] = true;
if ( stream_.doConvertBuffer[mode] )
@@ -4623,7 +4630,7 @@ void RtApiWasapi::wasapiThread()
RtAudioError::Type errorType = RtAudioError::DRIVER_ERROR;
// Attempt to assign "Pro Audio" characteristic to thread
HMODULE AvrtDll = LoadLibrary( "AVRT.dll" );
HMODULE AvrtDll = LoadLibrary( (LPCTSTR) "AVRT.dll" );
if ( AvrtDll ) {
DWORD taskIndex = 0;
TAvSetMmThreadCharacteristicsPtr AvSetMmThreadCharacteristicsPtr = ( TAvSetMmThreadCharacteristicsPtr ) GetProcAddress( AvrtDll, "AvSetMmThreadCharacteristicsW" );
@@ -4830,11 +4837,10 @@ void RtApiWasapi::wasapiThread()
stream_.deviceFormat[INPUT] );
if ( callbackPulled ) {
// Convert callback buffer to user sample rate and channel count
// Convert callback buffer to user sample rate
convertBufferWasapi( stream_.deviceBuffer,
convBuffer,
stream_.nDeviceChannels[INPUT],
stream_.nUserChannels[INPUT],
captureFormat->nSamplesPerSec,
stream_.sampleRate,
( unsigned int ) ( stream_.bufferSize * captureSrRatio ),
@@ -4848,7 +4854,7 @@ void RtApiWasapi::wasapiThread()
stream_.convertInfo[INPUT] );
}
else {
// no conversion, simple copy deviceBuffer to userBuffer
// no further conversion, simple copy deviceBuffer to userBuffer
memcpy( stream_.userBuffer[INPUT],
stream_.deviceBuffer,
stream_.bufferSize * stream_.nUserChannels[INPUT] * formatBytes( stream_.userFormat ) );
@@ -4924,35 +4930,27 @@ void RtApiWasapi::wasapiThread()
stream_.userBuffer[OUTPUT],
stream_.convertInfo[OUTPUT] );
// Convert callback buffer to stream sample rate and channel count
convertBufferWasapi( convBuffer,
stream_.deviceBuffer,
stream_.nUserChannels[OUTPUT],
stream_.nDeviceChannels[OUTPUT],
stream_.sampleRate,
renderFormat->nSamplesPerSec,
stream_.bufferSize,
convBufferSize,
stream_.deviceFormat[OUTPUT] );
}
else {
// Convert callback buffer to stream sample rate and channel count
convertBufferWasapi( convBuffer,
stream_.userBuffer[OUTPUT],
stream_.nUserChannels[OUTPUT],
stream_.nDeviceChannels[OUTPUT],
stream_.sampleRate,
renderFormat->nSamplesPerSec,
stream_.bufferSize,
convBufferSize,
stream_.deviceFormat[OUTPUT] );
}
// Convert callback buffer to stream sample rate
convertBufferWasapi( convBuffer,
stream_.deviceBuffer,
stream_.nDeviceChannels[OUTPUT],
stream_.sampleRate,
renderFormat->nSamplesPerSec,
stream_.bufferSize,
convBufferSize,
stream_.deviceFormat[OUTPUT] );
// Push callback buffer into outputBuffer
callbackPushed = renderBuffer.pushBuffer( convBuffer,
convBufferSize * stream_.nDeviceChannels[OUTPUT],
stream_.deviceFormat[OUTPUT] );
}
else {
// if there is no render stream, set callbackPushed flag
callbackPushed = true;
}
// Stream Capture
// ==============
@@ -4978,8 +4976,8 @@ void RtApiWasapi::wasapiThread()
if ( bufferFrameCount != 0 ) {
// Push capture buffer into inputBuffer
if ( captureBuffer.pushBuffer( ( char* ) streamBuffer,
bufferFrameCount * stream_.nDeviceChannels[INPUT],
stream_.deviceFormat[INPUT] ) )
bufferFrameCount * stream_.nDeviceChannels[INPUT],
stream_.deviceFormat[INPUT] ) )
{
// Release capture buffer
hr = captureClient->ReleaseBuffer( bufferFrameCount );
@@ -5047,8 +5045,8 @@ void RtApiWasapi::wasapiThread()
// Pull next buffer from outputBuffer
// Fill render buffer with next buffer
if ( renderBuffer.pullBuffer( ( char* ) streamBuffer,
bufferFrameCount * stream_.nDeviceChannels[OUTPUT],
stream_.deviceFormat[OUTPUT] ) )
bufferFrameCount * stream_.nDeviceChannels[OUTPUT],
stream_.deviceFormat[OUTPUT] ) )
{
// Release render buffer
hr = renderClient->ReleaseBuffer( bufferFrameCount, 0 );
@@ -5092,7 +5090,6 @@ Exit:
CoTaskMemFree( captureFormat );
CoTaskMemFree( renderFormat );
//delete convBuffer;
free ( convBuffer );
CoUninitialize();
@@ -6063,6 +6060,8 @@ void RtApiDs :: stopStream()
stream_.state = STREAM_STOPPED;
MUTEX_LOCK( &stream_.mutex );
// Stop the buffer and clear memory
LPDIRECTSOUNDBUFFER buffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0];
result = buffer->Stop();
@@ -6103,6 +6102,9 @@ void RtApiDs :: stopStream()
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!";
@@ -6136,6 +6138,8 @@ void RtApiDs :: stopStream()
unlock:
timeEndPeriod( 1 ); // revert to normal scheduler frequency on lesser windows.
MUTEX_UNLOCK( &stream_.mutex );
if ( FAILED( result ) ) error( RtAudioError::SYSTEM_ERROR );
}
@@ -6222,6 +6226,12 @@ void RtApiDs :: callbackEvent()
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] );
@@ -6402,11 +6412,12 @@ void RtApiDs :: callbackEvent()
}
nextWritePointer = ( nextWritePointer + bufferSize1 + bufferSize2 ) % dsBufferSize;
handle->bufferPointer[0] = nextWritePointer;
}
if ( handle->drainCounter ) {
handle->drainCounter++;
goto unlock;
}
// Don't bother draining input
if ( handle->drainCounter ) {
handle->drainCounter++;
goto unlock;
}
if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {
@@ -6545,6 +6556,7 @@ void RtApiDs :: callbackEvent()
}
unlock:
MUTEX_UNLOCK( &stream_.mutex );
RtApi::tickStreamTime();
}